# Why did Asterisk remove SIP in favor of PJSIP?

**URL:** <https://forums.vitalpbx.org/t/why-did-asterisk-remove-sip-in-favor-of-pjsip/3388>\
**Category:** FAQ\
**Created:** [October 26, 2023, 11:34am UTC](https://forums.vitalpbx.org/t/why-did-asterisk-remove-sip-in-favor-of-pjsip/3388 "2023-10-26T11:34:30Z")\
**Posts on this page:** 1\
**Page:** 1

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**Author:** ![admin](https://forums.vitalpbx.org/user_avatar/forums.vitalpbx.org/admin/32/3_2.png) [@admin](https://forums.vitalpbx.org/u/admin)\
**Post date:** [October 26, 2023, 11:34am UTC](https://forums.vitalpbx.org/t/why-did-asterisk-remove-sip-in-favor-of-pjsip/3388/1 "2023-10-26T11:34:30Z")

</div>

Starting with VitalPBX 4 we decided to completely remove SIP in favor of PJSIP, since this new technology provides better functionality.  
Also Asterisk decided to stop supporting SIP from version 17 and will remove it completely in version 21, which will be released very soon.  
Don’t worry, the VitalPBX Team has adapted the interface so that the transition from SIP to PJSIP is as user-friendly as possible. Everything you did in SIP, you can do with PJSIP.

We invite you to see the following link:

> **[Asterisk 21 Module Removal - chan\_sip ⋆ Asterisk](https://www.asterisk.org/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