[asterisk-users] asterisk pjsip webrtc rtp to private IP

marek cervajs64 at gmail.com
Thu Dec 12 04:38:45 CST 2019


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?


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