Hosterix Iran

Hosterix is wholly owned by Faraz Rasane Mihan Co. It was founded on January 2000. It started its activities by providing Web Services. In about 5 months after starting, Hosterix entered a new Category, which has had a rapid growth through the last years

Hosterix is wholly owned by Faraz Rasane Mihan Co. It was founded on January 2000. It started its activities by providing Web Services. In about 5 months after starting, Hosterix entered a new Category, which has had a rapid growth through the last years and is estimated to continue its growth over the next years; IP Telephony. By first providing PC-to-Phone services to our clients, now we are ready to provide Wholesale Services for Corporate, Enterprises and Service Providers. Right now we also have our own Calling Cards in Iran and decide to expand it to other countries in the near future.Networking and E-Commerce are newly added to Hosterix list of Services and we are seriously entering these markets.After all, we believe that IT is the core of today`s business and life. So, Hosterix will make itself involved!

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General office

Hosterix
Suite 5, 2nd Floor, #45, N.Eskandari St., Azadi Ave.
Tehran
Tehran ,14197
Iran

Phone: +98 216 913 055
Site: http://www.hosterix.com
Contact Person

Ramin Behzadi / CEO
Phone: +98 216 913 055
Phone: +98 911 200 389
  • International VoIP Wholesale Provider
  • Internet Telephony Service Provider
  • Voip Termination ISP
  • VoIP Billing Software Provider
  • Phone To Phone
  • Fax To Fax
  • PC to Phone
  • Termination
  • Billing Software
  • Partnering for Origination and Termination
  • H.323 Wireless/ GSM VoIP Solutions
  • Cisco Systems
  • Clarent
  • VocalTec
  • Lucent
  • Nuera
  • Vegastream
  • Huawei VoIP Products
  • None

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VoIP Favors SIP

VoIP Favors SIPSession 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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