[2017-06-13 04:41:50] WARNING[2982][C-00000004]: res_rtp_asterisk.c:773 ast_rtp_ice_start: No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)
– Channel SIP/100-0000000a joined ‘simple_bridge’ basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
– Channel SIP/600-00000009 joined ‘simple_bridge’ basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
[2017-06-13 04:42:21] NOTICE[1766]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call ‘SIP/600-00000009’ for lack of RTP activity in 31 seconds
– Channel SIP/600-00000009 left ‘simple_bridge’ basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
== Spawn extension (macro-dial-one, s, 51) exited non-zero on ‘SIP/600-00000009’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/600-00000009’ in macro ‘exten-vm’
== Spawn extension (ext-local, 100, 2) exited non-zero on ‘SIP/600-00000009’
– Executing [h@ext-local:1] Macro(“SIP/600-00000009”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/600-00000009”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/600-00000009”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/600-00000009”, “”) in new stack
If I call webRTC “res_rtp_asterisk.c:773 ast_rtp_ice_start: No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)” warning appeared and hangup . What can I do? I used freepbx version 13 and asterisk 13.12.Please help me.