SIPSRUS United States

SIPSRUS is a SIP Trunk provider cuts down actual cost of communication within and outside the business infrastructure by 50 percent on inbound and outbound phone service giving you the stellar opportunity to reduce costs effectively whilst maintaining eff

As we grow, we are guided by the following values:

To be passionate about everything we do.

To listen to our customers and provide them with quality solutions, that is the best value for their money and which brings a smile to their face.

To maintain our integrity and nurture an open, innovative and fun open source work culture.

To be a leader, leading a team to success, exceeding expectations in the assigned tasks, or taking new initiatives that create value for the company and the customer.

To Innovate, newer ideas to change the game, the processes, the engineering.

We expect our customers to take a second and view how we can indulge in your service. Feel free to write us if you have queries regarding the technologies offered. You can email us at info@sipsrus.com

Thank you

SIPSRUS Management

Your rate:
Rating: 0/0.00/5

General office

SIPSRUS
79 Pine Street, Suite 235
New York City
New York ,10005
United States

Phone: 877.388.1971
Site: http://www.sipsrus.com/
Contact Person

James Smith / Project Manager
Phone: 877.388.1971
Not data
  • Click To Call
  • Asterisk
Not data

Providers in database: 6433
Register VoIP Provider

RTP Delivers Multimedia Data In Real Time

RTP Delivers Multimedia Data In Real TimeA 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.

View more WIKI
Austria DIMAWEB Network Solutions e.U.

Ihr Hosting- u Telefonprovider sowie EDV-Dienstleister aus Österreich.

Read more