菜鸟学freeswitch（二）webRTC拨软电话自动挂断

2019-01-08 17:39:49.221806 [ERR] mod_sofia.c:2343 CODEC NEGOTIATION ERROR. SDP:
v=0
o=- 3481799438673358832 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS MP9KUnaULVHFpwTqhoSfv1fphnZ1jZqMmqsP
m=audio 57630 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 192.168.1.19
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2382557538 1 udp 2113937151 192.168.1.19 57630 typ host generation 0 network-cost 999
a=ice-ufrag:B3vK
a=ice-pwd:V7sBbym5dh+buN6G60VzS1zZ
a=ice-options:trickle
a=fingerprint:sha-256 01:E0:C0:8A:2A:6D:3A:C1:89:51:01:1D:49:A1:BB:93:7F:BF:09:0D:20:18:1A:FB:5A:32:B0:35:71:FF:79:FB
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=rtcp-mux
a=rtcp-fb:111 transport-cc
a=ssrc:1707582762 cname:IEUyTgVxrZhvH1Wf
a=ssrc:1707582762 msid:MP9KUnaULVHFpwTqhoSfv1fphnZ1jZqMmqsP 8824eac9-c860-4baa-bcd6-ae06a2103a4f
a=ssrc:1707582762 mslabel:MP9KUnaULVHFpwTqhoSfv1fphnZ1jZqMmqsP
a=ssrc:1707582762 label:8824eac9-c860-4baa-bcd6-ae06a2103a4f

这种问题是webRTC默认的编码问题

 

 

 修改internal.xml

在xml的最后添加

<param name="apply-candidate-acl" value="rfc1918.auto"/>
<param name="apply-candidate-acl" value="localnet.auto"/>　　

 

最后按F6 reloadxml  再去试试 应该就好了 再不行重启freeswitch

 

转载于:https://www.cnblogs.com/pangyangqi/p/10240351.html