PortaSIP is a call control software package enabling service providers to build scalable, reliable VoIP networks. Based on the Session Initiation Protocol (SIP), PortaSIP provides a full array of call routing capabilities to maximize performance for both small and large packet voice networks.
PortaSIP allows IP Telephony Service Providers (ITSPs) to deliver communication services at unusually low initial and operating costs that cannot be matched by yesterday’s circuit-switched and narrowband service provider PSTN networks.
In addition to conventional IP telephony services, PortaSIP provides a solution to the NAT traversal problem and enhances ITSP network management capabilities. It can be used to provide residential, business (cloud PBX), and wholesale traffic exchange services. PortaSIP provides the following functionalities:
- SIP registration, allowing SIP phones to use the service from any IP address (static or dynamically assigned).
- Multiple hosted PBX environments on the same physical server.
- Real-time authorization for all calls, limit on the maximum number of simultaneous calls per customer.
- NAT traversal, media proxying, protection against DoS (Denial of Service) attacks.
- Multi-lingual (voice) error announcements from the media server and customizable greeting upon successful service activation.
- Automatic disconnect of calls when the maximum credit time is reached; ability to dynamically lock the funds required to cover the next interval, thus ensuring overdraft protection even if multiple calls are made concurrently.
- Automatic disconnect of calls when one of the parties goes offline due to a network outage.
- Various PBX features: call waiting, call transfer, call hold, music on hold, hunt groups, follow-me, etc.
- Fail-over routing – a list of routes arranged according to cost, preference, and customer routing plan is supplied by PortaBilling.
- Forwarding of calls to the voicemail service (Media Server) if a SIP phone is not available.