Sachitel Communications India

Sachitel is a VoIP and telecommunications services provider focused on the Call Center and Teleservice market space. We currently provide high quality, cost-effective telecommunications services to call centers and teleservice centers all over the world, including, the continental US, Europe, India and Pakistan.

Sachitel is a leading provider of VoIP termination and long distance termination to the US, UK and Canada. We offer a full product suite for origination from the US including toll free numbers , and local numbers (DIDs) in all major markets in the US, UK and Canada. We offer both Inter-Asterisk? Exchange (IAX and IAX2) termination and Session Initiation Protocol (SIP) termination. We also support Asterisk? the open-source Linux PBX.

Our rates are among the best in the industry. Allow us to show you why we are the fastest growing VoIP and telecommunications provider to the Call Center and Teleservices industry! Call or contact us for more information.

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General office

Sachitel Communications
A -38 K , Sector 64
Noida
U.P. ,201301
India

Phone: 911202404413
Phone: 911202404414
Site: http://www.sachitel.com

Contact Person

Rahul Sharma / Manager
Phone: 919818702735
Phone: 919215687770

International VoIP Wholesale Provider
Internet Telephony Service Provider
Hosted VoIP billing service provider
Internet VoIP and Video Conferencing Service Provider
Voip Termination ISP
VoIP Billing Software Provider
Voip consulting
SIP Billing

Phone To Phone
Phone To PC
Web Call
Outsourced Billing
PC to Phone
vFone
IP devices
Web To Phone
Collaboration
Data Conferencing
Voice and Video Conferencing
Billing Software
Call Relay VoIP Solutions
Call Routing VoIP Solutions
Partnering for Origination and Termination
Consultancy Services
Installation and Support Services
H.323 Wireless/ GSM VoIP Solutions
SIP Softswitch & CPE
SIP VoIP Gateway

Cisco Systems
Mera
Audiocodes

ProfitBilling
PortaSIP

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Providers in database: 6433

VoIP Favors SIP

Session Initiation Protocol (SIP) is a text-based standard put in place by the Internet Engineering Task Force (IETF) and is the primary protocol for Voice over Internet Protocol services. SIP is similar to a MGCP in that it's a signaling protocol that can create, maintain and terminate sessions in IP-based networks. These sessions include multimedia conferencing and two-way phone calls. SIP protocol can run on User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) and Transmission Control Protocol (TCP) because it was created to function independently from the underlying Transport Layer as an Application Layer.

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