VOIP Softphones
The softphone works like a telephone to let user make calls using the desktop computer. The user can make calls to anyone via the internet who also has a softphone installed. Further if user have signed to a broadband phone company or to a VOIP IP-PBX then he can connect to the VOIP network and can make calls to the regular telephone numbers.
MARS services include the development and customization of softphone applications for various platforms. Some of the features the MARS has developed for Softphone are:
- Configure phone to support upto 5 lines with ability to put calls on hold
- User name, Caller ID and Time display
- Phone book with search facility based on name and number
- Speed dial support
- Buttons for various services like Call hold, Call transfer, Conference, Do Not Disturb etc. Configurable buttons to provide user specific settings
- Configure phone to work behind NAT and firewalls.
- Automatic pop ups for the various problems like headset/mike is not connected or system is behind firewall etc.
- Includes data compression (GSM, uLaw, ALaw, PCM and G726), echo cancellation, noise reduction, comfort noise and more.
- Can play music on-hold to the callers.
- Support Instant Messaging with the users having the same softphone.
- Customize phone to support two widely used VOIP protocols SIP and H.323 so that the phone can connect to any of the H.323 or SIP based IP-PBX.
MARS has deep expertise to work with technologies such as:
- Microsoft .Net framework
- J2EE
- development on Windows, Windows Mobile (PocketPC PDAs and SmartPhones), Linux operating systems
- SIP/H.323 based call control
- NAT and NAPT traversal on any kind of network using STUN, TURN, ALG
- LDAP, radius protocols
- Voice codecs G.729, G.723, G.722, G.711, GSM
VOIP Protocols
Voice-Over-IP (VOIP) implementations enable users to carry voice traffic (for example, telephone calls and faxes) over an IP network. VOIP system consists of number of different components like, Gateway/Media Gateway, Gatekeeper, Call Manager, and Messaging Server. The IP Network is used for two types of communications, one is signaling and other is the voice media.
MARS offers a suite of outsourcing service offerings - new product development, test services, porting services and sustenance and support - for VOIP service providers. MARS is expertise in VOIP protocols like SIP, H.323 and MGCP/MEGACO.
Session Initiation Protocol
The Session Initiation Protocol (SIP) is a signaling, presence and instant messaging protocol developed to set up, modify, and tear down multimedia sessions, request and deliver presence and instant messages over the Internet. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution.
SIP is IETF protocol for VOIP, the latest specification of the SIP is defined in RFC3261.
SIP is client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services like call hold, call forward, call transfer, conferencing, Automatic Call Distribution, Caller and Callee authentication etc.
H.323
The H.323 standard provides a foundation for multimedia communications across IP-based networks, including the Internet. H.323 is an "umbrella" specification, which includes the standards H.323, H.225.0, H.245, the H.450-series documents, and the H.460-series.
H.225.0 defines the call signaling between endpoints and the Gatekeeper. The communication with the Gatekeeper establishes the Registration, Admission and Status (RAS) sessions with endpoints. H.245 signaling is used to negotiate capabilities and to control aspects of the call between two or more endpoints.
H.323 supports advanced audio and video conferencing and provides standard mechanism for variety of services including Call transfer, Call forward, Call par/pickup, Call hold, Call waiting and MWI etc.
Media Gateway Control Protocol
Media Gateway Control Protocol (MGCP/MEGACO) is used for controlling media gateways from external call control elements called media gateway controllers (MGC) or call agents (CA). A media gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.
MGCP removes the signaling control from the media gateway by putting it into the MGC/CA. In the MGCP architecture, the intelligence (control) is unbundled from the media (data). It is a master-slave protocol where the master has absolute control and the slave simply executes commands. The master is the MGC, or CA and the slave is the media gateway
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