[asterisk-dev] One sip stack to rule them all....

[Gunnar Hellström]

The Real-Time Text feature of Asterisk does not work with PJSIP. Or at 
least it is not documented how its redundant transport support is 
configured.
See: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4260034 
for how it once worked.

(There are bugs in the release 11 and 13 implementations of redundant 
transmission of real-time text with chan_sip as well, but it worked in 
earlier releases. )



Den 2017-10-08 kl. 16:55, skrev James Finstrom:
> One does not simply depricate a sip stack.
>
> Ok so at devcon there was a discussion of depricating chan_sip. This 
> may sound a lot worse than it actually is. Chan_sip has been 
> essentially untouched in 4ish years. It does not receive bug fixes. It 
> is just sort of a barge floating in the ocean.
>
> So one of the things that is needed to finally put Chan sip to bed is 
> feature parody.  Someone brought up CCSS.
>
> What features do you feel you would lose going from chan_sip to pjsip.
>
> Are there any bugs in pjsip that keep you from migrating?
>
>

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