Open realJoshByrnes opened 11 months ago
I had look at porting the patch to chan_pjsip, however because the features there are split into separate modules it wouldn't work as for example; the phones expect subscriptions to be working during register.
The easiest option would be to port chan_sip to Asterisk 21 but remove any parts that the phones don't need.
To strip back chan_sip it would probably be better to rename the transport to something like chan_ucm (or similar)
I've only been using chan_sip for Cisco devices for quite some time now. I'm guessing that would be the same for most people with Cisco devices.
Are you or anyone else stepping up to maintain chan_sip in Asterisk 22+? Can you expand more on why it would not be possible to port to pjsip? Is it a matter of being a lot of work and no ability/time/motivation to do it, or an actual limitation in the pjsip architecture that makes it impossible to support Cisco's phones?
chan_pjsip is implements SIP as a set of separate modules that may or may not be loaded. It is possible to support registering endpoints but not subscriptions or mailboxes etc, also some of the request/response handling is done via generic callbacks that don't support endpoint specific behaviour.
Cisco phones need almost all of SIP to be available and in the cases where Cisco-specific extensions are required those callbacks will need to be modified to include an extra parameter.
That would take additional extra work over what it took for chan_sip and it would never be accepted upstream, at that point it would be better to create a new Cisco specific SIP channel driver that also uses libpjsip.
Just wondering what the future holds for these patches after Asterisk 20?
It's clear from the codebase of Asterisk 21 that chan_sip is removed.
I guess the options would be to:
Have you put any thought into this?