iNextrix Technologies India
iNextrix Technologies specializing in telecom Solutions, has come to be one of the industry''s leading solutions provider companies. iNextrix Technologies has been providing total solutions to our customers, creating innovative solution is our expertise.
iNextrix is a Asterisk/ VoIP service provider, we are leaders in providing turnkey solutions in Asterisk
* Asterisk: IVR/IP PBX/SOFT SWITCH
* Asterisk Gateway Interface (AGI)
* End to End VOIP Set-up
* Predictive Dialer
* Call Centre Solutions
* Softphone
* SER/billing soft switch
* LCR/CDR
* Web to call
* Click to call
* Inter-office/Intra-office Intercom
* IP Phones
* IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices
* Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream
iNextrix Technologies
Ahmedabad
Ahmedabad
Gujarat ,380014
India
Phone: +91 70 40307595
Site: http://www.inextrix.com
- FREE Callshop Software
- Hosted PBX Service Provider
- Hosted VoIP billing service provider
- Mobile VoIP
- SIP Billing
- VoIP Billing Software Provider
- Voip consulting
- Billing Software
- Call Back
- Call Routing VoIP Solutions
- Call shop solution
- Consultancy Services
- Data Conferencing
- E1/T1
- Gatekeepering
- gsm voip solutions
- H.323 VoIP Gateway
- Hosted PBX
- Hosted VoIP
- Installation and Support Services
- IP devices
- Outsourced Billing
- PC to Phone
- Phone To PC
- Phone To Phone
- Polycom
- Project Mangement Services
- SIP VoIP Gateway
- System Integration
- Termination
- Virtual PBX
- Voice and Video Conferencing
- VoipSwitch
- Web Call
- Web To Phone
- Asterisk
- Asterisk
- Asterisk
- Audiocodes
- D-LINK
- Digium
- grandstream
- Huawei VoIP Products
- Linksys
- Quintum
- SIP softphones
- a2billing
- another
- Billing system of own development
- Custom Open Source Billing Solution
- IP PBX
- IP Phone
- Voip Switch
A Real-time Transport Protocol (RTP) is a protocol standard defined by the Internet Engineering Task Force (IETF) that outlines a management system for programs using real-time transmission of data through mutlicast or unicast network services in multimedia situations. Originally designed by the IETF for videoconferencing for multiple participants, the protocol is largely used in Voice over Internet Protocol applications. Despite its name, the protocol cannot guarantee real-time delivery of data, however, the RTP does compensate for jitters and can detect when data arrives out of sequence, both issues being popular in VoIP communication. In IP telephony, RTP works with a signaling protocol, such as SIP or H.323, in order to set up connections in a network.
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