Asterisk 21 Module Removal – chan_sip

chan_sip will no longer be included with Asterisk as of the release of version 21.

Deprecated in version 17, chan_sip has been scheduled for removal for some time.

The replacement, chan_pjsip has been in production on countless systems for a number of years.  Sangoma’s own PBXs switched years ago and many providers now require or at least strongly encourage it’s use when connecting to their services.

If you haven’t become familiar with chan_pjsip yet, the best time was yesterday but today will have to do.  chan_pjsip is named so because it uses pjsip to handle the SIP messaging.  pjsip itself is not maintained by the Asterisk team.  We do however, work closely with its maintainers to resolve issues that may potentially affect Asterisk.

This has worked out well so far and ensures that both Asterisk and the SIP stack that it uses are vetted and well maintained.  This helps us be responsive to issues and feature requests while also making sure that the SIP stack used by Asterisk remains efficient and secure.  Testing now only needs to be done against one SIP stack, simplifying our build and test processes.

Within the Asterisk release is a script to help with this conversion, – but please refer to the res_pjsip wiki to make sure that the configuration options post conversion are appropriate for your system.  A good starting point can also be found here.  chan_pjsip also includes support for configuration wizards, which can simplify endpoint configuration greatly.

chan_sip is not disappearing from the internet.  We understand that there are those who are not yet willing to part with chan_sip.  The patch(es) for it’s removal will be generated as part of the removal process.  One of the beautiful things about open source software is that you can fork and maintain for yourself whatever you are able.  Other current versions of Asterisk will continue to include chan_sip

chan_pjsip is the future, and also as of Asterisk 21 – the present.


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