Kamailio
Kamailio is an open source SIP server used for VoIP and real-time communications. It handles call routing, policy enforcement, media handling, accounting, and authorization with support for presence and instant messaging.
sip
voip
realtime-communications
call-routing
policy-enforcement
media-handling
accounting
authorization
presence
instant-messaging
Kamailio: Open Source SIP Server for VoIP and Real-Time Communications
Kamailio is an open source SIP server used for VoIP and real-time communications, handling call routing, policy enforcement, media handling, accounting, and authorization with support for presence and instant messaging.
What is Kamailio?
Kamailio is an open source Session Initiation Protocol (SIP) server used for Voice over IP (VoIP) and real-time communications applications and services. It handles fundamental SIP protocol needs like call routing, policy enforcement, media handling, accounting, and authorization, with support for additional functions like presence, instant messaging, and more.
Some key features and capabilities of Kamailio include:
- Routing calls and messages between SIP endpoints like IP phones and SIP clients
- NAT traversal for connecting devices behind NAT routers
- Policy control for call admission, quality of service, and access rules
- Media proxying for managing voice, video, and instant messaging media sessions
- Accounting and billing data generation for call detail records
- Authentication and authorization using protocols like RADIUS and LDAP
- Extensive support for presence services and instant messaging
- High performance and scalability using techniques like async communication
Kamailio is used to build large SIP infrastructure for carriers, telecom providers, VoIP service providers, and enterprises. It can enable applications like IP PBX systems, VoIP providers, SIP trunking, routing servers, and real-time communication solutions. The open source nature promotes customization, flexibility, and cost-effectiveness.
Kamailio Features
Features
- SIP proxy and registrar server
- Presence and instant messaging
- Load balancing and failover
- NAT traversal
- Media relaying
- Accounting and billing
- Flexible routing and policy enforcement
- Modular architecture with extensive API
Pricing
- Open Source
Pros
Open source and free
Highly scalable and performant
Rich feature set for VoIP
Active community support
Modular and customizable
Supports wide range of SIP devices and clients
Cons
Steep learning curve
Requires expertise to setup and manage
Not as user friendly as commercial options
Limited official support and documentation
Official Links
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