31 May
2024
31 May
'24
8:59 a.m.
On May 31, 2024, at 8:35 AM, Alex Balashov via
sr-users <sr-users(a)lists.kamailio.org> wrote:
Thanks Alex for the high praise. =)
On May 31, 2024, at 8:09 AM, Benoît Panizzon via
sr-users <sr-users(a)lists.kamailio.org> wrote:
This would also solve another problem. We have some CPE with limited
memory which struggle with a long record-route list.
Indeed, and this is why I recommended it. It would solve a larger class of problems,
above and beyond the one you currently face with RTPEngine per se.
-- Alex
But with Asterisk I would have to add quite some
config to pass on required additional customer sip header.
Yeah, Asterisk also doesn’t allow for media bypass from the start. Instead it requires the
pbx to anchor media and then does a re-invite to bypass; which is just not a great method
(imo). With FreeSWITCH not doing any media it is quite performative and also allows for
the call to be “taken over” by freeswitch with it’s standard uuid controls.
Regards,
Fred Posner
p: +1 (352) 664-3733
https://fred.tel