On May 31, 2024, at 8:35 AM, Alex Balashov via sr-users sr-users@lists.kamailio.org wrote:
On May 31, 2024, at 8:09 AM, Benoît Panizzon via sr-users sr-users@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
Thanks Alex for the high praise. =)
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