1TM Guatemala
Expertos en Linux y telefonia IP Asterisk.
Marcas:
VIOPBX, Sangoma, Digium, Polycom, Cisco, Linksys, Astra, Bria, Eyebeam, y mas.
General office
1TM
12 calle 12-76 zona 10 Colonia Oakland
Guatemala
Guatemala ,01010
Guatemala
Phone: +50223296280
Site: http://www.i-t-m.com
Contact Person
Manuel Andrade / Gerente Comercial
Phone: +50223296280 EXT.400
1TM
12 calle 12-76 zona 10 Colonia Oakland
Guatemala
Guatemala ,01010
Guatemala
Phone: +50223296280
Site: http://www.i-t-m.com
Contact Person
Manuel Andrade / Gerente Comercial
Phone: +50223296280 EXT.400
- Cloud-based Unified Communications Service Provider
- did numbers provider
- FREE Callshop Software
- Hardware
- Hosted PBX Service Provider
- Hosted VoIP billing service provider
- Internet Fax Service Provider
- Internet Telephony Service Provider
- Internet VoIP and Video Conferencing Service Provider
- Marketing Services
- Mobile VoIP
- Network Service Provider
- SIP Billing
- VoIP Billing Software Provider
- Voip consulting
- Voip Termination ISP
- Wholesale VoIP Carrier
- Consultancy Services
- Data Conferencing
- E1 Bulk Wholeseller
- E1/T1
- Fax To Fax
- Gatekeepering
- gsm voip solutions
- Hosted PBX
- Hosted VoIP
- Installation and Support Services
- IP devices
- Marketing Services
- Outsourced Billing
- Partnering for Origination and Termination
- PC to Phone
- PC to Phone, Phone To PC, Phone To Phone, Project Mangement Services, Termination, Web Call, Web To Phone
- Phone To PC
- Phone To Phone
- Polycom
- Project Mangement Services
- SIP Softswitch & CPE
- Sip Trunking
- SIP VoIP Gateway
- System Integration
- Virtual PBX
- Voice and Video Conferencing
- Web Call
- Web To Phone
- 2N
- Asterisk
- Asterisk
- Asterisk
- Audiocodes
- D-LINK
- Digium
- grandstream
- Linksys
- SIP softphones
Not data
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.
Impro Labs LLC
Voice Over IP (VOIP) communications have became the major trend in international calling. As the IP communication is done over internet, good internet connectivity with high bandwidth is required to ensure good quality of service. This is where our Optima
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