Asterisk PJSIP Support in Adhearsion Core

[sfgeorge]

Summary

Asterisk's chan_sip SIP driver is deprecated in favor of chan_pjsip. Adhearsion should be updated to support both drivers.

In an ideal world, Adhearsion could automatically detect which driver incoming calls are based on (based on agi_type: "SIP" or "PJSIP"). But outbound calls at minimum, we're going to need to introduce a configuration option to allow one to choose between the two.

While Asterisk allows a mixed-mode in which it can handle some calls on chan_sip with other calls on chan_pjsip at the same time – I propose that we do not support co-operating sip/pjsip modes in Adhearsion for the sake of API simplicity. Rather, I suggest a single global sip/pjsip option in Adhearsion Configuration.

What Needs to Change

I've spotted only a few places that need to change to be able to support PJSIP.

• Setting special SIPADDHEADER... variables is a chan_sip-specific construct. We need to update Originate actions to pre-set SIP headers with PJSIP when appropriate.
• Asterisk PJSIP Support in adhearsion-asterisk adhearsion-asterisk#19

Related Docs

• https://mojolingo.com/blog/2015/adhearsion-compatible-asterisk-13/
• https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
• https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels

[sfgeorge]

There's a slight chance we may be able to avoid adding a new configuration value, even when originating brand-new calls. We'd have to dynamically inject a special [context] into the asterisk dialplan for injecting the SIP headers at the last minute, at which point we could examine ${TECH} to see if it is "SIP" or "PJSIP".

This would mimic an approach that FreePBX has taken:

1. SIP header(s) are added to a ${SIPHEADERS} channel variable (Hash) as needed.
   https://github.com/FreePBX/core/blob/release/15.0/Dialplan/dialparties.php#L953
2. They've dynamically injected a [func-apply-sipheaders] context (a Gosub) into the dialplan which adds the headers for the appropriate channel driver depending on whether ${TECH} is "SIP" or "PJSIP".
3. When invoking Dial(), they include b(func-apply-sipheaders^s^1) in order to invoke that Gosub on the newly created channel and apply the SIP headers.