VERIMOR TELEKOM Turkey
Verimor Telecommunication Services established in 2008 May for giving services customers and achieved the telecom operator license in 2009 june as a STH operator. We are providing to customers network solutions , telecommunication services, and system int
Local and licensed operator in Turkey with VERIMOR TELECOM cheap and high quality sound to make VoIP termination, you want?
SUMMARY OF TARIFF
Country Price Per Minute
TURKEY, FIX, IN AREA 0,0154 $ USD
TURKEY, FIX, OUT AREA 0,0205 $ USD
TURKEY, FIX, TRNC 0,0463 $ USD
TURKEY, FIX, 444 SPECIAL SERV. 0,0405 $ USD
TURKEY, MOBILE, TURKCELL 0,0287 $ USD
TURKEY, MOBILE, TURKCELL KKTC 0,1820 $ USD
TURKEY, MOBILE, VODAFONE 0,0296 $ USD
TURKEY, MOBILE, VODAFONE KKTC 0,1956 $ USD
TURKEY, MOBILE, AVEA 0,0339 $ USD
SERVICE FEATURES
- Full CLI support
- SIP v2 and H.323
- G711, G723 and G729 codec support
- 100% termination in Turkey networks
- One second pricing
- Unlimited Channel support
- DTMF support (RFC2833)
- 7/24 NOC e-mail support
- Through web panel, Cdr Reports, balance, online payment
- Compatible with all softswitchs
- Best Pricing
- Top quality sound
Omer CURGUL
Project Coordinator
omer@verimor.com.tr
Please! Contact my E-Mail Address
VERIMOR TELEKOM
TALATPASA MAH. ASLANGAZI CAD. AKISLI SOK. NO:3 D:4 OKMEYDANI
ISTANBUL
KAGITHANE ,34000
Turkey
Phone: +902123205062
Site: http://www.verimor.com.tr
Contact Person
OMER CURGUL / PROJECT COORDINATOR
Phone: +905338190482
- Hosted VoIP billing service provider
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- Internet Telephony Service Provider
- Voip Termination ISP
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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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