We provide telecommunications network infrastructure
Reference in telephone system
Boost your company's communication
SIPPulse content and articles
Product demonstration
Videos and manuals
Currently, Sippulse works on three product offering fronts:
A software platform for measuring voice over IP services, designed for operators and voice service providers — it features an advanced SIP signaling structure capable of serving subscribers quickly and reliably.
Its main characteristics are flexibility, scalability and integration capacity, making it capable of supporting private voice and media networks, whether for companies or government organizations with large geographic distribution and a high number of employees.
Softswitch is based on the SIP protocol and operates through a highly sophisticated engine that implements the rules defined by the ITU (International Telecommunication Union), also supporting the RFC's defined by the IETF (Internet Engineering Task Force).
The SIPPulse SBC is a SIP gateway that groups 4 functions in its configuration.
Its main function is to represent the voice over data network element provided for in the ITU specifications for NGN networks, responsible for the security and demarcation of networks in IP telephony (SBC).
The SBC acts both at the exit of the network and at the entrance network as a proxy, to authenticate users connected via public networks, or the internet. It also acts as a B2B agent to connect with other providers through a SIP Trunk — in these functions, the SBC does not manipulate or interfere with dialing or signaling introduced by the user.
It can be configured as a SIP-i gateway, for SIP interconnections with encapsulated SS7 signaling, and can act as a WEBrtc gateway, receiving signaling from applications built on compatible browsers.
A solution aimed at implementing telephone services on private networks.
The SIPPulse HPBX platform offers an integrated communication services environment for companies with high volume, complexity and dispersion of internal communication with their community of customers and suppliers.
It can be offered on a dedicated basis to user companies, and to ISP providers, or offered to offer cloud telephony to its corporate subscribers.
It includes services for administrative use (PABX) and Customer Service (SAC), installed together, in addition to URAV (URA with visual management) and Dialer (automated approach to customers).
Our pricing model is based on the number of simultaneous calls and number of subscribers.
This will depend on the type of operation and the volume of information that will be transmitted in the environment. The systems can be installed in a physical environment — whether or not your own data center — and a virtualized environment, such as Amazon, Google, Oracle, etc.
Although our software has complete dashboards that make it easier to monitor your operation, SIPPulse encourages you to develop your own dashboards to monitor the systems, based on your specific needs. For this, we recommend using Zabbix.
The platform generates daily backups during the night — the time may vary depending on each client's traffic period. The backup is stored for 3 days, after this period it is deleted.
Therefore, we recommend that the backup be stored on another server for future consultation, especially for STFC operators, considering that Anatel requires that call data be stored for 5 years.
Currently all SIP systems work on top of Centos 7.
In this case, it is recommended to contact our specialized support as quickly as possible, considering that this type of situation can paralyze your operation.
To prevent this from happening, our software has a login banner that monitors the status of the partitions — starting from version 5.0, a field was added to the dashboard that shows in real time how the disk is occupied.
Furthermore, we recommend that alerts be created via zabbix to assist in quick decision-making in case of increased disk usage.
It is each SIP packet that is exchanged during a phone call. This data is stored in the database and used to create the CDR.
CDR is the call ticket. It is created based on the duration between an INVITE ACC and a BYE ACC.
Unfortunately, our Softswitch does not include this function, but we developed BCORE, which in addition to controlling the franchise, also controls its numbering base.
For STFC customers, it is possible to enter data relating to the NSAPN via the interface. To do this, you need to access the STFC → Cadup → enter the data relating to portability.
Processing calls are calls that are still exchanging SIP packets but have not been established. In the case of active calls, these are those that have already been answered and are in conversation.