Hello All, I am using Asterisk 11.4 to make WebRTC calls. I am having one way audio issues in certain scenarios.
Analysis Failure case: -------------------------------- INVITE sent from browser: v=0^M o=- 546074467646554532 2 IN IP4 127.0.0.1^M s=-^M t=0 0^M a=group:BUNDLE audio^M a=msid-semantic: WMS 9oVZAJxsBQKy0FFy9FyYvp3OlN4xNHHknuuC^M m=audio 33746 RTP/SAVPF 111 103 104 0 8 106 105 13 126^M c=IN IP4 194.183.244.5^M a=rtcp:33746 IN IP4 194.183.244.5^M a=candidate:1679965505 1 udp 2113937151 10.1.5.116 50461 typ host generation 0^M a=candidate:1679965505 2 udp 2113937151 10.1.5.116 50461 typ host generation 0^M a=candidate:2999745851 1 udp 2113937151 192.168.56.1 42208 typ host generation 0^M a=candidate:2999745851 2 udp 2113937151 192.168.56.1 42208 typ host generation 0^M a=candidate:3890964107 1 udp 2113937151 10.1.65.38 47247 typ host generation 0^M a=candidate:3890964107 2 udp 2113937151 10.1.65.38 47247 typ host generation 0^M a=candidate:2265168813 1 udp 1845501695 194.183.244.5 33746 typ srflx raddr 10.1.5.116 rport 50461 generation 0^M a=candidate:2265168813 2 udp 1845501695 194.183.244.5 33746 typ srflx raddr 10.1.5.116 rport 50461 generation 0^M a=candidate:715243953 1 tcp 1509957375 10.1.5.116 0 typ host generation 0^M a=candidate:715243953 2 tcp 1509957375 10.1.5.116 0 typ host generation 0^M a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0^M a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0^M a=candidate:2842204795 1 tcp 1509957375 10.1.65.38 0 typ host generation 0^M a=candidate:2842204795 2 tcp 1509957375 10.1.65.38 0 typ host generation 0^M a=ice-ufrag:olXlAxnuMOs4Lro/^M a=ice-pwd:5XHaFtt6AnMTGzGEH838T+vP^M a=ice-options:google-ice^M a=fingerprint:sha-256 2A:89:A9:D9:08:1B:56:1F:68:91:51:46:98:02:95:38:65:C3:F2:6E:DC:FD:F5:7D:C2:BD:8F:D9:4B:CC:39:61^M a=setup:actpass^M a=mid:audio^M a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level^M a=sendrecv^M a=rtcp-mux^M a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:jY2eb+Kf9rWU8RrD0c0A/MEef/M15jAiTMkx/XaZ^M a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vj3FFZX3DEIKpS+FcHgp89aqPuHvSGdMEICgKaeQ^M a=rtpmap:111 opus/48000/2^M a=fmtp:111 minptime=10^M a=rtpmap:103 ISAC/16000^M a=rtpmap:104 ISAC/32000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:106 CN/32000^M a=rtpmap:105 CN/16000^M a=rtpmap:13 CN/8000^M a=rtpmap:126 telephone-event/8000 a=maxptime:60^M a=ssrc:2744681183 cname:A3AoKceVBFkjBBi0^M a=ssrc:2744681183 msid:9oVZAJxsBQKy0FFy9FyYvp3OlN4xNHHknuuC 9oVZAJxsBQKy0FFy9FyYvp3OlN4xNHHknuuCa0^M a=ssrc:2744681183 mslabel:9oVZAJxsBQKy0FFy9FyYvp3OlN4xNHHknuuC^M a=ssrc:2744681183 label:9oVZAJxsBQKy0FFy9FyYvp3OlN4xNHHknuuCa0 Asterisk 2xx response :: v=0^M o=root 972456278 972456278 IN IP4 81.201.82.213^M s=Inum^M c=IN IP4 81.201.82.213^M t=0 0^M m=audio 12256 RTP/SAVPF 8 0 126^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:126 telephone-event/8000^M a=fmtp:126 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=ice-ufrag:2a1217a53e86c40c66e149bd57f769d6^M a=ice-pwd:6eb2a7476f494b160d39df395af10d01^M a=candidate:H51c952d5 1 UDP 2130706431 x.x.x.x 12256 typ host^M a=candidate:H51c952d5 2 UDP 2130706430 x.x.x.x 12257 typ host^M a=connection:new^M a=setup:active^M a=fingerprint:SHA-256 F2:F6:3F:50:01:EF:89:B3:D5:8C:9B:D2:A9:FA:3A:5B:61:0C:67:E1:8B:AA:65:4C:A0:14:45:49:BE:F0:42:69^M a=sendrecv^M Call is connected, but i observed in chrome://webrtc-internals and saw that packets received by the browser is "ZERO". Analysing it further with a PCAP trace, i realized that asterisk is STUN binding request to one of the host candidates (which is a private IP) proposed in the incoming INVITE. It doesn't attmept other candidates. Interesting observation if i disable one of my local interface created by Oracle VM virtual box(192.168.56.1), everything is fine. ------ I would really appreciate if someone could throw light on this issue and help me debug it. Thanks, Nitesh Bansal
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