Now there is a call scenario where freeswitch ->kamailio ->rtpengine ->webrtc
(jsSIP) voice and video cannot be displayed. Is there a problem with my
handling and how can I solve it?
1、Freeswitch INVITE to kamailio without video code
2、kamailio request to rtpengine(rtpengine_offer("record-call=o
May I ask what the problem is? It prompts an error but does not affect the
program's operation. Is it that the configuration file and the deployed version
are inconsistent?
13(682) ERROR: *** cfgtrace:dbg_cfg_trace(): branch_route=[MANAGE_BRANCH]
c=[/usr/local/etc/kamailio/kamailio.cfg] l=706 a
