hi,
i have following topology
PSTN - Asterisk ---- internet ----- router - jssip client (wss)
Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP
connection to PSTN
router - public IP/private IP (NAT)
jssip client - private IP - sip over websocket to Asterisk PJSIP
~30% of calls has problem with no audio. reason is that Asterisk is
sending RTP to private IP of jssip
SDP looks the same for good call and bad call too
i searched through res_rtp_asterisk.c but i'm not sure where to put
DEBUG info about which IP and why Asterisk pick for RTP
any hint?
is it possible debug Asterisk STUN request/response ? or is it hidden in
pjsip internals?
Marek
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