a2zTEL Communications Inc Canada
A2ZTEL is a global leader in delivering high-quality international and domestic long distance calling services to subscribers at a fraction of the cost of traditional carriers by leveraging the power of the Internet.
A2ZTEL is a cost effective commun
A2ZTEL is a global leader in delivering high-quality international and domestic long distance calling services to subscribers at a fraction of the cost of traditional carriers by leveraging the power of the Internet.
A2ZTEL is a cost effective communications service provider that is an alternative to traditional telephony services. A2ZTEL's products and services are characterized by value, ease-of-use and superior technology. Regardless of the internet connection, whether dial-up or high speed, and combined with our software and hardware devices, we offer innovative, feature-rich, digital voice quality telephony services.
A2ZTEL adopts the Session Initiation Protocol (SIP) which is the most advanced Voice over Internet Protocol (VoIP).
A2ZTEL has the right solution for all sizes of business.
a2zTEL Communications Inc
1255 Boul Laird
Montreal
Quebec ,H3P-2T1
Canada
Phone: 514-807-5163
Site: http://www.a2ztel.ca
Contact Person
Sales a2ztel / Sales
Phone: 514-807-5163
- Hardware
- Hosted VoIP billing service provider
- International VoIP Wholesale Provider
- Internet Telephony Service Provider
- Network Service Provider
- SIP Billing
- Voip consulting
- Billing Software
- Call Routing VoIP Solutions
- Consultancy Services
- Gatekeepering
- H.323 Softswitch & CPE
- H.323 VoIP Gateway
- H.323 Wireless/ GSM VoIP Solutions
- Outsourced Billing
- PC to Phone
- Phone To PC
- Phone To Phone
- SIP Softswitch & CPE
- SIP VoIP Gateway
- Web Call
- Cisco Systems
- Clarent
- Lucent
- Mera
- NACT
- Quintum
- Billing system of own development
- Oracle Based own system
- PortaSIP
- ProfitBilling
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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