From this change (res_rtp_asterisk): ast 13.10 to 13.11 webrtc JSSIP stop working, failing withchan_sip.c:4083 retrans_pkt: Hanging up call7238b48c11581d4166b899bf747a05f7@130.211.62.184:0 – no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions)..