The best open-source WebRTC SIP server is likely to be a matter of personal preference and the specific requirements of your project. Some popular options include:
Kamailio: A high-performance SIP server with support for WebRTC and many other protocols. It is widely used in large, carrier-grade deployments.
Jitsi: A set of open-source projects that provide secure video conferencing and instant messaging capabilities. It includes the Jitsi Meet WebRTC-based video conferencing platform, as well as the Jitsi SIP Phone.
Asterisk: A popular open-source PBX and telephony platform that can be used to build WebRTC applications using the PJSIP SIP stack and the res_rtp_asterisk module.
FreeSWITCH: A versatile telephony platform that can be used to build a wide variety of applications, including WebRTC-based solutions.
It’s best to evaluate each one and see which one fits your needs.
Kamailio
Kamailio is a high-performance SIP (Session Initiation Protocol) server that is widely used in large, carrier-grade deployments. It is an open-source project that was first released in 2005 and is actively maintained by a community of developers.
One of the main strengths of Kamailio is its high performance and scalability. It is able to handle a large number of simultaneous SIP connections and can easily handle traffic from hundreds of thousands of users. Kamailio also supports many advanced features such as load balancing, failover, and traffic shaping, making it suitable for use in high-availability environments.
In addition to its core SIP capabilities, Kamailio also supports a wide range of other protocols, such as WebRTC, RTP, RTCP, TCP, and UDP. This makes it a versatile platform that can be used to build a wide variety of telephony and real-time communication applications.
Kamailio also has a modular architecture, which allows developers to easily add new features and functionalities using Kamailio modules. There are many community-developed modules available, covering various features such as presence, instant messaging, and more.
Kamailio is also well-documented, and there is a large community that provides support and assistance to users. There are also many commercial companies that provide professional support and consulting services for Kamailio.
Overall, Kamailio is a powerful and flexible SIP server that is well-suited for use in large-scale, high-availability environments and it can be used in a wide range of telephony and real-time communication applications.
Jitsi
Jitsi is a set of open-source projects that provide secure video conferencing and instant messaging capabilities. It includes the Jitsi Meet WebRTC-based video conferencing platform, as well as the Jitsi SIP Phone.
Jitsi Meet is a web-based conferencing platform that allows users to participate in video, audio, and text chats. It uses WebRTC for real-time communication and is built on top of the Jitsi Videobridge for multi-party video conferencing. Jitsi Meet also provides end-to-end encryption for secure communication. It can be easily installed on a server or used as a cloud-based service.
Jitsi SIP Phone is a softphone application that allows users to make and receive phone calls using SIP (Session Initiation Protocol). It is built on top of the SIP.js library and provides a simple, easy-to-use interface for making and receiving calls. It can be used with a variety of SIP servers, including Kamailio and Asterisk.
The Jitsi projects are built using open standards and protocols, which makes them highly interoperable with other systems. Jitsi Meet can be easily integrated into other applications, such as chat clients and web portals, and it can also be used in conjunction with other communication tools, such as email and instant messaging.
Jitsi is also well-documented and has a large community of users and developers. The Jitsi team also provides commercial support and consulting services for Jitsi Meet.
Overall, Jitsi is a set of powerful and flexible open-source projects that provide secure video conferencing and instant messaging capabilities. Jitsi Meet is a web-based conferencing platform that allows users to participate in video, audio, and text chats, Jitsi SIP Phone is a softphone application that allows users to make and receive phone calls using SIP.
Asterisk
Asterisk is a popular open-source PBX (Private Branch Exchange) and telephony platform that can be used to build a wide variety of communication applications, including those that use WebRTC.
Asterisk is built on top of a highly configurable and customizable architecture, which allows developers to add new features and functionalities using Asterisk modules. It supports a wide range of communication protocols, including SIP (Session Initiation Protocol), IAX (Inter-Asterisk eXchange), H.323, MGCP (Media Gateway Control Protocol), and SCCP (Skinny Client Control Protocol).
Asterisk also supports WebRTC by using the PJSIP SIP stack and the res_rtp_asterisk module. This enables developers to easily integrate WebRTC functionality into their applications and allows users to make and receive voice and video calls directly from web browsers.
Asterisk is widely used in many different industries, from small and medium-sized businesses to large enterprise deployments. It can be used to build a wide range of telephony applications, such as PBX systems, voicemail systems, call centers, and conferencing systems. It can also be used to build custom communication applications, such as IVR (Interactive Voice Response) systems, and can be integrated with other software, such as CRM (Customer Relationship Management) systems.
Asterisk has a large community of users and developers and is well-documented. There are also many commercial companies that provide support and consulting services for Asterisk.
Overall, Asterisk is a powerful and flexible open-source platform that can be used to build a wide variety of communication applications, including those that use WebRTC. It’s modular architecture and support for a wide range of protocols make it suitable for a variety of industries and use cases.
FreeSWITCH
FreeSWITCH is a versatile open-source telephony platform that can be used to build a wide variety of communication applications, including those that use WebRTC.
FreeSWITCH is built on a modular architecture, which allows developers to easily add new features and functionalities using FreeSWITCH modules. It supports a wide range of communication protocols, including SIP (Session Initiation Protocol), IAX (Inter-Asterisk eXchange), H.323, MGCP (Media Gateway Control Protocol), and SCCP (Skinny Client Control Protocol). It also supports WebRTC through its mod_sofia and mod_verto modules.
FreeSWITCH is highly configurable and customizable, allowing developers to tailor it to their specific needs and requirements. It can be used to build a wide range of telephony applications, such as PBX systems, voicemail systems, call centers, and conferencing systems. It can also be used to build custom communication applications, such as IVR (Interactive Voice Response) systems, and can be integrated with other software, such as CRM (Customer Relationship Management) systems.
FreeSWITCH is widely used in many different industries, from small and medium-sized businesses to large enterprise deployments. It is also used in many carrier-grade deployments. The FreeSWITCH community is large and active, and there are many commercial companies that provide support and consulting services for FreeSWITCH.
Overall, FreeSWITCH is a powerful and flexible open-source telephony platform that can be used to build a wide variety of communication applications, including those that use WebRTC. Its modular architecture and support for a wide range of protocols make it suitable for a variety of industries and use cases. It’s a widely used and well-documented platform, and has a large community of users and developers that provide support and assistance.