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Kamailio as the SIP server

Kamailio is an open source SIP server under GPLv2+. Its page lists proxy, registrar, WebSocket, and scaling in front of a PBX.

Learn in 5 minutes

  1. Quote the role from the Kamailio home page
  2. Name WebSocket support for WebRTC as a listed feature
  3. Decide when the SIP server is Kamailio and the dialplan is elsewhere

Visual

Edge proxy in front of a PBX · study drawing · cyphermonkeylabs.com BrowserKamailioAsterisk INVITE INVITE 180 180

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This is the page's list, not a module index. Open the documentation page before you enable a module.

Worked example

Who answers the dialplan Kamailio received the INVITE. Asterisk is the next hop.

  1. Read the two roles.
  2. Say where extensions.conf is.
  3. Leave RTP to rtpengine if you add it.

Fault

Kamailio routes the INVITE. The caller expects voicemail prompts. Nothing in the SIP server is configured as that PBX feature.

The home page separates the SIP server from the PBX and media server it can scale.

Quiz

Kamailio is released under


The scale phrase on that page is


The page says Kamailio can front


WebSocket support for WebRTC is


Least cost routing and load balancing are


The dialplan for an Asterisk behind Kamailio still lives in


Ecosystem

Glossary, sister systems, and the integration seam for this room.

Analytics Amazon Connect can attach conversational analytics to an external voice system with a SIPREC connector. The analytics plane is not a second SIP proxy for the customer call.

Glossary · Ecosystem map · Ports · Troubleshooting

Sources and go deeper