VoIP Telephony Server
Our VoIP Telephony Server is based on SIP protocol and compatible with many softphones and HW VoIP-enabled phones. SIP Proxy, Registrar and Redirect Server in one, with NAT Traversal for operation behind firewalls and routers. Connectivity to external PBX Gateways to route calls across carrier's networks, based on custom call-forwarding and number rewriting rulesets.
Lower Telecommunication Costs You don't need to know anything about SIP protocol to cut down the company's telephone bill. Simplified configuration, one-click deployment, error-free operation, no special settings required on client side make it the ideal solution for company of any size.
>Well-suited for sales departments
>Follows SIP principles
>All calls ring to all users registered for the number
>The call is rejected only when all users are offline
>The one who first picks up wins
>Great flexibility with number rewriting, presence and internal chat support
SIP Registrar, Proxy and Redirect Server in one
Parent SIP Proxy mode is intended for distributed SIP servers, interconnection with existing servers/appliances and to bridge separate local network subnets, so that the call doesn't have to go through an external router/firewall, but instead locally through the proxy. Operation of Registrar and Redirect server is fully automated, based on access rights, number rewrites and forwarding rules created by administrator.
>Proxy serves as a bridge between SIP servers or separate networks
>Registrar assigns VoIP number to group of users
>Redirect interconnects to external PBX Gateways
>Forward calls to mobile networks, carrier transatlantic VoIP gateways with QoS,..
>Route calls to other IceWarp VoIP Servers in different location
Operation behind firewalls and routers
Error-proof system for routing calls across firewalled networks designed to be fully transparent for user agents, meaning that no special network configuration is required on either phone endpoint.
>SDP NAT Traversal (Port Proxy)
>Smart on-the-fly IP-IP matching system, no connectivity problems
>Fully transparent for user agents, connection feels like peer-to-peer
>No configuration required on client side
>Automatically assigned IP address in server configuration
Your Own Secretary
Dial from any contact list (Outlook, WebMail, IM client) and your phone will ring when the call is already placed. Works with any phone, no driver required, no proprietary communication protocols between device and application. Call is established by the server and redirected to your default VoIP number.
>Dial from Outlook contact list via Oulook Connector plug-in
>Pickup the phone when it rings the other party
>Set your available/away state by a quick number dial
>Redial last called number Dial the last caller number
>Get complete call log sent to you by email
>Use your favourite HW phone as a headset
>No need to change phones or install any proprietary software
>Ultimate compatibility, wireless convenience w/o dedicated headsets
>Dial email addresses on phones with numeric keypad
Your Own Calling Plan
Free calls to local and remote VoIP users. Just 1 public VoIP number is necessary for simultaneous calls to carrier networks and conferences. By engaging a Voice-over-IP gateway or router to PBX, you can enable your FXO analog telephones for Internet calls without additional wiring or change of equipment.
>Audio codec (G.711, iLBC, GSM EFR,..) independent
> Wide choice of free/commercial computer softphones
> Standalone WiFi or Ethernet SIP IP phones
> Dual mode mobile handsets such as Nokia N95, E61
> VoIP gateways for FXO devices
> Use SIP for local or VoIP enabled, call via carrier gateway or to PBX
Your Own Numbering Plan
Route calls as needed through very flexible, prefix based rules. Assign external numbers to email account IDs. Custom Quick dials to dial Last called/Last incoming, Set/Reset Away status with possibility of call forwarding. Rules can be chained for great flexibility and wide range of usage scenarios unlike HW SIP solutions. No RegEx knowledge is required with GUI interface and intuitive syntax with wildcard convention (%).
Ultimate call control features Extensible call forwarding, call redirection and number rewrite rulesets Least cost routing to gateways Call forwarding (to another local/external number) Separate settings for incoming and outgoing calls Rewrite outgoing number to pretend to be a different number Rewrite incoming number to have it assigned to a defined group of users Manage all routed connections, add new routes dynamically
> Authentication through existing account
> Server access or restricted to internal users only
> Allowed hosts, subnets, wildcards
> Call-logging to user mailbox and/or global file
> User permissions to use specific gateways
Future Technologies Today
Support for extended DNS lookup allows to discover the right communication channel and to reach the user through SIP if possible. ENUM schema (Electronic Numbering) can be viewed as an indirect dialling service designed to seamlessly interconnect PSTN and VoIP. It maps conventional telephone numbers (E164 numbering system) to Internet addressing, by the use of NAPTR extended DNS records. Based on SRV records, the correct communication channel will be decided accordingly, so there can be the same domain name for mail, SIP, FTP.
SIP SIMPLE instant messaging and presence capabilities allow peer-to-peer or server based chat conversations.
> SIP SIMPLE instant messaging
> ENUM schema support
> NAPTR SRV extended DNS records look-up
> Detailed logs per user, can be parsed for billing
> All calls logging
> Restrictions for calls going through a gateway
> Unlimited gateways
> API for extensions and system integration
> Integrates with other subsystems (Mail, IM, GW)- endless possibilites
> Feature open, cooperation with VoIP vendors welcome
> 4 tabs, unlimited possibilities
> Simple setup-users authenticate with their existing mail account
> Automatic NAT setting system
> Define/modify rules on fly
> User selection dialogs
Integration Benefits
> One place authentication, no need to install, interconnect or configure > No middleware,fully transparent for user agents > Same authentication box, policy, authentication schema, trusted hosts > Familiar GUI configuration, logs, service settings > Fast setup, universal, simple for anybodyThe GXC mission is to establish, develop and deliver value added management and technical consultancy to companies involved in mobile and fixed satellite communication networks, addressing all aspects system development and deployment lifecycle.
