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VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine

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Kamailio cfgs

kamailio configurations and associated files for various usecases and role that kamailio can fit into

Usecases

  • Barebonde SIP Server - use bare minimum modules and replies to any incoming call , no relay or proxy , no nat etc
  • REGISTER handle - just replies 300 ok to every REGISTER type request method
  • Stateful transaction handle
  • Simple Relay - relays incoming SIP messages to another end
  • Record routing
  • NAT handle
  • REGISTER and Userloc
  • RTPENgine media proxy
  • Accounting
  • WebRTC WS SIP Server

Note : Used kamailio v5.x , many old examples and sample configs from older wiki sources have been updated here too

TLS protocol method

Possible values are:

  • TLSv1.2 - only TLSv1.2 connections are accepted (available starting with openssl/libssl v1.0.1e)
  • TLSv1.1+ - TLSv1.1 or newer (TLSv1.2, ...) connections are accepted (available starting with openssl/libssl v1.0.1)
  • TLSv1.1 - only TLSv1.1 connections are accepted (available starting with openssl/libssl v1.0.1)
  • TLSv1+ - TLSv1.0 or newer (TLSv1.1, TLSv1.2, ...) connections are accepted.
  • TLSv1 - only TLSv1 (TLSv1.0) connections are accepted. This is the default value.
  • SSLv3 - only SSLv3 connections are accepted. Note: you shouldn't use SSLv3 for anything which should be secure.
  • SSLv2 - only SSLv2 connections, for old clients. Note: you shouldn't use SSLv2 for anything which should be secure. Newer versions of libssl don't include support for it anymore.
  • SSLv23 - any of the SSLv2, SSLv3 and TLSv1 or newer methods will be accepted.

Ref : https://downloads2.goautodial.org/files/version4/etc/kamailio/kamailio-wss+sip.cfg