What is a SIP Server?
SIP stands for session initiation protocol. A SIP server is a network protocol that is used for establishing connections for communication of different subscribers and also deals with call management. Also, SIP servers are often used to manage call connection in VoIP solutions. A SIP server can
- Set up a connection between multiple endpoints
- Initialize media parameters for the endpoint, using SDP protocol
- Modify and adjust the parameters during the session
- Replace one endpoint with an another or new endpoint
- Session termination
Difference between PBX and SIP-enabled PBX
A PBX is a system that connects the individual extensions to the external mobile networks or telephone lines, whereas a SIP-based PBX connects to the internet and allows to make calls over the internet using SIP protocol.
Different VoIP SIP servers and their GUI
Here in this article let us understand various VoIP SIP servers and their User Interfaces that are used to manage each SIP server.
Asterisk
Asterisk is an interactive voice response platform that includes an automatic call distributor functionality. It is an open-source PBX that allows building own communication applications. It is a framework that is used for building real-time communication solutions and multi-protocol solutions. Digium sponsors it.
Features of the asterisk
- Call monitoring
- Call transfer
- Call waiting
- Append Message
- Blind transfer
Functions
- Small server footprint for processing and memory capabilities
- It has Longlife and support mechanisms
- It allows access to gateway configurations. Documentation, forums, etc
- For auto-provisioning it has an extensive database of end device templates.
Protocols
- ISDN
- SS7
- MGCP
- H.323
- IAX
- SIP
Supported OS
- Linux
- BSD
- Solaris
- Mac OS X
GUI- The Graphical User Interface that is used to manage the Asterisk is Free PBX
Free switch
It offers the most flexible to build its unified communication suite. It is majorly focused on
- Cross-platform support
- Stability
- Modulatory
- Scalability
Freeswitch as the essential calling features and also includes the advanced features like PSTN interfaces for digital and analog circuits, speech recognition and synthesis.
The free switch makes use of the freely available software libraries that allows the required functions of the system; this way, it reduces the complexity of the system.
Functions
- It has a Multi-tenant platform, and each tenant is segregated
- Freeswitch has clustering capabilities
- Concurrent calls, when given the same underlying hardware, can be maximized with increased capabilities.
Protocols
- STUN
- SCCP
- MRCP
- SIP
- IAX
- Skype
- ISDN
- H.323
- JINGLE
- SIMPLE
- XMPP
- RSS
Supported OS
- Mac OS X
- Linux/BSD
- Windows
- Solaris
GUI- The Graphical User Interface that is used to manage the Freeswitch is FusionPBX
Yate
The abbreviation of Yate is Yet another telephony Engine. Yate is an extensible GPL licensed PBX and open-source communication software that supports
- Video
- Voice
- Instant messaging
It supports scripting languages like PHP, Unix shell, Python, Perl even though C++ is its core software.
Main components of Yate
- Core
- Message engine
- Telephony engine
- Yate modules
Protocols
- H.323
- SIP
- MGCP
- SCTP
- MAP CAMEL
- SS7 over IP
- IAX
- ISDN
- JINGLE
- XMPP
- Cisco SLT
- SCCP
- TCAP
Supported OS
- Mac OS
- BSD
- Windows
- Linux
GUI- The Graphical User Interface that is used to manage the Yate is Yateclient
Elastix
Elastix has support for a wide range of hardware like Digium, Snom, Yealink. It has included a call center module with a predictive dialer. It only offers up to 8 free SIM calls to 25 users. Elastix offers unified communications server software, which includes the following features:
Features of Elastix
- Instant Messaging
- Faxing
- Collaboration functionality
- Integrated softphones for Mac and Windows
Elastix also includes the features that are brought from other open-source projects like Postfix, HylaFax, FreePBX, Openfire
Kamailio/ OpenSER
Kamailio, previously known as OpenSER, is a free and open-source sip sever and offers a high-security level. Compared to other SIP servers, Kamailio is a bit difficult to adopt as it requires deep knowledge of the SIP protocol. To provide secure communications, it has powerful features like
Features of Kamailio
- SCTP
- TLS
- Asynchronous TCP, UDP
- Instant messaging
- Least cost routing
- Load balancing
- Routing fail-over
- Authentication and authorization
Due to its ability to provide high-level encryption, it is one among the top secured servers and is recommended to businesses that prefer the security and wants everything to be kept inside.
GUI- The Graphical User Interface that is used to manage the Kamailio is Siremis
OpenSIPs
OpenSIPs is one of the fastest SIP servers that offer robust and scalable solutions at an enterprise level. It is a multi-functionality sip server that majorly targets delivering a high-level technical solution which can be used in professional SIP server platforms. This technical Solution providers mainly includes
- Quality
- Performance
- Security
Features of OpenSIPs
- Multi-domain support
- Perl programming Interface
- Least cost routing
- Variables support in the script
- IPv4 and IPv6
- Modular architecture
- Call processing language
GUI- The Graphical User Interface that is used to manage the OpenSIPs is OpenSIPs CP
Flexisip Server
Flexisip is a scalable and modular SIP server that offers all the required to deploy an own SIP service for desktop or mobile applications. Integrating Flexisip into your SIP infrastructure is easy, and it serves various purposes effectively.
Features of Flexisip
- Real-time statistics through a command-line interface
- Push notifications
- Group chats
- Real-time presence status
- Identifying users of service within the address book
It is essential to understand and do proper research before choosing the SIP server for your VoIP Solution setup. A single server can not meet all your needs, and every SIP server has its pros and cons. You wisely need to choose the VoIP SIP server that meets most of your requirements.
Krify is a leading VoIP service providing company with competency in customizing Linphone softphone by using Fusion PBX. We have customized softphones for various organizations and have developers who are expertise in Linphone development including the customization with advanced features. For more information reach us here.
