Hi, I am at the point where connection is established and no apparent errors are reported.
However audio is not output. The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160) Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160) Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160) And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143) What could be causing there to be no audio? This is the connected sdp =0 o=root 350315728 350315728 IN IP4 10.31.xxx.xxx s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.31.xxx.xxx t=0 0 m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2 a=ice-pwd:12611b8146599a9019d59b4b649a7970 a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
