The softswitch part is the main element of the platform, which merges the functionality of the following VoIP architecture elements:
- SIP registrar
- SIP proxy
- SMS gateway
- SIP (RFC 3261)
- SMS through SIP, HTTP and SMPP
- Fax through T38
The main characteristics of the softswitch include:
- Various types of proxy methods e.g. full proxy (with RTP-proxy), signaling proxy and other options, possibility of selecting a proxy method per destination, route or per client.
- Full interoperability with industry standards compatible VoIP equipment (gateways, switches, ATAs, terminals).
- Bidirectional NAT support for SIP equipment.
- Advanced routing system (support for internal virtual prefixes that allows the creation of separate dialing plans for different groups of customer accounts).
- Routing based on prefixes, priorities per routes, depending on allowed voice Codecs per destination.
- Video calling
- Support for failover (rerouting), configurable end reasons initiating fail over, support for priorities.
- Load sharing support – Advanced algorithm taking care of traffic being evenly distributed according to defined percentages for multiple routes.
- Least Cost Routing (LCR).
- Internal numbering plans support.
Various authentication methods:
- By IP address
- By caller ID (ANI)
- By SIP credentials
Flexible methods for call setup data modifications (for clients and/or for destination in the dialing plan):
- Modifying dialed number, adding prefixes/suffixes, wild cards, max/min number length.
- Modifying caller ID/SIP display – Adding prefixes/suffixes, wild cards, max/min number length.
- Defining allowed and/or primary Codecs for clients and for terminators.
- Codecs auto negotiation.
- Import/Export accounts and dialing plan from/to excel or TXT file.
- Settings stored in the MySQL database.
- Scalability is supported due to a cluster configuration, where multiple VoipSwitch servers run connected to each others in what is known as “cluster”, sharing the same SQL database server, thus increasing performance by dividing the traffic among the multiple servers while retaining a central point of management with one main IP address for clients (load balancing).
- Redundancy support for seamless traffic handover in case of the main server failure, the service allows for controlling availability of particular ports (for example SIP listeners) real-time SQL data backup procedure.
- Music on hold
- Do not disturb
- Call transfer: blind and attended
- Follow me/Find me (based on caller ID of incoming call), sequential or ring to all
- Voicemail boxes with personalized voice greetings for different caller IDs
- Hunt/Ring groups
- Voicemail to email with attachment (Mp3)
- Voicemail notification to SMS or email
- Voicemail transcription to SMS
- SMS forwarding (e.g. internal SIP SMS forwarding to external GSM numbers)
- T38 support
- Email to fax
- Fax to email
- Fax inbox – Viewing received faxes from the Portal
- Sending faxes directly from the Portal
- support for graphic files, pdf and doc/docx
- SIP SIMPLE protocol
- Support for Notify/Publish methods
- Roster (buddies) management on the server side
- Presence support built-in the mobile and PC softphones
- Presence in the Portal