TekSIP

TekSIP

KaplanSoft
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About

Kamailio® (successor of former OpenSER and SER) is an open-source SIP server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications, presence, WebRTC, Instant messaging, and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems, or media servers like Asterisk™, FreeSWITCH™, or SEMS. Among the powerful features, are asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; simple instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached.

About

TekSIP is a SIP Registrar and SIP Proxy for Windows. TekSIP supports UDP, TCP, TLS and WebSocket (IPv4 & IPv6) transports. TLS and Secure WebSocket are supported in only commercial editions. TekSIP is tested on Microsoft Windows Vista, Windows 7/8/10/11 and Windows 2008-2022 server. TekSIP can be deployed as a signaling server for WebRTC based SIP phones. TekSIP complies with RFC 3261, RFC 3263, RFC 3311, RFC 3581 and RFC 3891. It supports NAT traversal and ENUM. Please see technical details about NAT traversal in Readme. You can select IP address to be listened and alternative SIP endpoints for outgoing calls. You can also log session details into a log file and monitor active registrations and sessions in real-time. You can monitor number of active registrations and sessions using Windows Performance Monitor (Perfmon.exe). TekSIP can redirect calls to alternate endpoint when called endpoint is unavailable (Off-line, busy...). Active SIP session can be terminated via GUI.

Platforms Supported

Windows
Mac
Linux
Cloud
On-Premises
iPhone
iPad
Android
Chromebook

Platforms Supported

Windows
Mac
Linux
Cloud
On-Premises
iPhone
iPad
Android
Chromebook

Audience

Companies and professionals in need of a tool to develop secure and extensible SIP servers to improve their VoIP services

Audience

Enterprise, Service Providers, Utility Companies

Support

Phone Support
24/7 Live Support
Online

Support

Phone Support
24/7 Live Support
Online

API

Offers API

API

Offers API

Screenshots and Videos

Screenshots and Videos

Pricing

No information available.
Free Version
Free Trial

Pricing

$239
Free Version
Free Trial

Reviews/Ratings

Overall 0.0 / 5
ease 0.0 / 5
features 0.0 / 5
design 0.0 / 5
support 0.0 / 5

This software hasn't been reviewed yet. Be the first to provide a review:

Review this Software

Reviews/Ratings

Overall 0.0 / 5
ease 0.0 / 5
features 0.0 / 5
design 0.0 / 5
support 0.0 / 5

This software hasn't been reviewed yet. Be the first to provide a review:

Review this Software

Training

Documentation
Webinars
Live Online
In Person

Training

Documentation
Webinars
Live Online
In Person

Company Information

Kamailio
Founded: 2001
United States
www.kamailio.org/w/

Company Information

KaplanSoft
Founded: 2011
Turkey
www.kaplansoft.com

Alternatives

Asterisk

Asterisk

Sangoma Technologies

Alternatives

TekIVR

TekIVR

KaplanSoft
VaxVoIP

VaxVoIP

VaxSoft
websockets

websockets

Python Software Foundation

Categories

Categories

Telephony Features

Auto-Dialer
Call Center Management
Call Monitoring
Contact Management
Inbound Reporting
IVR / Voice Recognition
Outbound Reporting
Predictive Dialer
Telemarketing Management
Voice & Data Integration
VoIP

Integrations

360 Monitoring
Apache Cassandra
Asterisk
Fonimo
FreeSWITCH
LDAP
MongoDB
MySQL
Oracle Cloud Infrastructure
PostgreSQL
Radius
Redis
TekRADIUS
memcached

Integrations

360 Monitoring
Apache Cassandra
Asterisk
Fonimo
FreeSWITCH
LDAP
MongoDB
MySQL
Oracle Cloud Infrastructure
PostgreSQL
Radius
Redis
TekRADIUS
memcached
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