Caryn Systems Russia

CARYN SYSTEMS OU was founded in Estonia in 2004 as a division of the telecommunication holding Telix Ltd., innovator in the telecom market since 1991.

CARYN SYSTEMS OU provides wholesale A-to-Z VoIP termination with high quality and competitive rates. High TDM quality is provided by QoS software and reliable Tier1 vendors. Our rate policy is designed to keep prices low, while maintaining a high level of quality.

Our customers will get individual payment terms. Post-payment is possible and personal rate sheets can be prepared, if necessary.

Our own network capacity is more than 10000 concurrent calls. We support H323 and SIP protocols, ISDN-PRI and SS7 connection is also possible.

CARYN SYSTEMS OU is always interested in new vendors who provide high-qualified termination with competitive rates. Currently, we have over 25 million VoIP minutes originated monthly into networks of our vendors.

We require only high-qualified termination of our VoIP traffic. All our vendors should recognize that our QoS system will automatically remove them from routing if consistent bad quality of service is detected. We always inform our vendors about all problems faced during usage of their service.

CARYN SYSTEMS OU offers consulting service for VoIP carriers and our own researched software solutions for telecom and other business branches. We can provide you with software according to your unique needs.

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General office

Caryn Systems
St. Petersburg Russia
St. Petersburg
,993434
Russia

Phone: 8(812)702-66-05
Site: http://www.caryn.ee

Contact Person

Ralston Willoughby / Purchase Maanger
Phone: 8(812)7026605
Phone: 8(905)2235356

Internet VoIP and Video Conferencing Service Provider

H.323 Softswitch & CPE
H.323 VoIP Gateway
H.323 Wireless/ GSM VoIP Solutions
PC to Phone
Phone To PC
Phone To Phone
Web Call
Web To Phone

Cisco Systems

another

Slavtelecom

MKTC

Providers in database: 6433

RTP Delivers Multimedia Data In Real Time

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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