Average Ratings 0 Ratings
Average Ratings 0 Ratings
Description
Kamailio® is an open-source SIP server, which evolved from the earlier OpenSER and SER, and is distributed under GPLv2+, capable of managing thousands of call setups each second. This robust platform can be utilized for constructing extensive systems for VoIP, real-time communication, presence, WebRTC, instant messaging, and various other applications. Additionally, Kamailio is well-suited for scaling SIP-to-PSTN gateways, PBX systems, or media servers such as Asterisk™, FreeSWITCH™, or SEMS with ease. Its extensive feature set includes support for asynchronous TCP, UDP, and SCTP, secure communication through TLS for voice, video, and text, as well as WebSocket compatibility for WebRTC. The server supports both IPv4 and IPv6, offers simple instant messaging and presence capabilities with an integrated XCAP server and MSRP relay, and enables asynchronous operations. Furthermore, it includes IMS extensions for VoLTE, ENUM functionality, DID and least cost routing, load balancing, and routing failover. Kamailio also provides comprehensive accounting, authentication, and authorization features, while supporting a variety of backend systems, including MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached. All these capabilities make Kamailio a versatile choice for modern communication infrastructures.
Description
A robust and adaptable software architecture allows for seamless integration of our services into current systems. Our offerings include VoIP, video, presence, and chat functionalities that can be accessed from any desk phone, softphone, IP PBX, or mobile/WebRTC device. Utilizing state-of-the-art technology that relies on standard APIs and interfaces, we are well-prepared for the advancements of Web 3.0 with our RTC engine technology. By significantly reducing the total cost of ownership, both operational and capital expenses, you can enhance your profit margins. Our platforms are utilized by leading ISPs across Europe, catering to hundreds of thousands of users each. The Sipwise Class 5 serves as a highly scalable SIP Softswitch that meets the diverse requirements of residential and business voice solutions right out of the box. Its modular design allows for easy expansion, as users can simply activate various software modules that are included in the initial setup, thereby ensuring adaptability and growth. This flexibility ensures that the system can evolve with changing demands and technologies.
API Access
Has API
API Access
Has API
Integrations
360 Monitoring
Apache Cassandra
Asterisk
Fonimo
FreeSWITCH
LDAP
MongoDB
MySQL
Oracle Cloud Infrastructure
PostgreSQL
Integrations
360 Monitoring
Apache Cassandra
Asterisk
Fonimo
FreeSWITCH
LDAP
MongoDB
MySQL
Oracle Cloud Infrastructure
PostgreSQL
Pricing Details
No price information available.
Free Trial
Free Version
Pricing Details
No price information available.
Free Trial
Free Version
Deployment
Web-Based
On-Premises
iPhone App
iPad App
Android App
Windows
Mac
Linux
Chromebook
Deployment
Web-Based
On-Premises
iPhone App
iPad App
Android App
Windows
Mac
Linux
Chromebook
Customer Support
Business Hours
Live Rep (24/7)
Online Support
Customer Support
Business Hours
Live Rep (24/7)
Online Support
Types of Training
Training Docs
Webinars
Live Training (Online)
In Person
Types of Training
Training Docs
Webinars
Live Training (Online)
In Person
Vendor Details
Company Name
Kamailio
Founded
2001
Country
United States
Website
www.kamailio.org/w/
Vendor Details
Company Name
Sipwise
Founded
2008
Country
Austria
Website
www.sipwise.com
Product Features
Product Features
Telephony
Auto-Dialer
Call Center Management
Call Monitoring
Contact Management
IVR / Voice Recognition
Inbound Reporting
Outbound Reporting
Predictive Dialer
Telemarketing Management
VoIP
Voice & Data Integration