Apelby Communications Vietnam
We are a Germany-based wholesale carrier with offices in the Czech Republic, Slovenia, Malaysia and Vietnam. We specialize in doing VoIP service since our establishment in 2005 and we have continued to grow in both the number of customers and amount of minutes.
Currently we are interconnected with around 200 carriers specially many Tier-1 carriers as T-Mobile, Telefonica, BT, Vodafone, Verizon, Orange, China Mobile, Bharti Airtel, Tata Communications, BICS... and we are terminating around 150 million voice minutes monthly.
Dear Partner,
Greeting from Apelby Communications. We are a Germany-based wholesale carrier with offices in the Czech Republic, Slovenia, Malaysia and Vietnam. Currently we are terminating 150 million minutes monthly and we are now promoting VOIP Vietnam Direct route with the following features:
- Connection: Voice IP with OPEN RTP. We have direct interconnection to the Mobile Network Operators in Vietnam The bandwidth is 15xE1 with each operator.
- Route quality: ASR: 65% - 75%; ACD 8 -10 mins, subject to traffic source (wholesale or retail), CLI provided. We will open RTP so you can check the IP connection of the operators.
- Rate: competitive rate base on volume projection.
- Prepayment term.
- Billing: 1+1 increment, 30-day period
Please feel free to contact me for more discussion about policies as well as rate.
Skype : duyduc01
Email: pham@apelby.com
Thank you.
Best regards,
Pham Duy Duc
Apelby Communications
Floor 19, Building 1, Trung Yen. Cau Giay dist.,
Hanoi
,100000
Vietnam
Phone: +84 462602720
Phone: +84909064078
Site: http://www.apelby.com
Contact Person
Pham Duy Duc / Carrier Relations Manager
Phone: +84 462602720
Phone: +849 09064078
- A-Z Termination
- Termination
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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