Re: [Freeswitch-users] No audio for conference call
Search your log for 'Choose rtp'; the IP number following this shows what FS will use for a media stream. There is a line like this for each stream. For your scenario, you want these not to be private IPs. If they are private, you can try to auto-reject them by changing acl.conf.xml. But you'll have limited success due to this bug https://github.com/signalwire/freeswitch/issues/157
Hi David Search your log for 'Choose rtp'. Did you mean fs_cli log? at fs_cli log, i didn't find anything 'Choose rtp'. However, siptrace log is as follows send 883 bytes to udp/[103.50.168.50]:46849 at 07:23:03.166391: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.103.30:46849 ;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport=46849;received=103.50.168.50 From: <sip:2112@52.221.122.187;transport=UDP>;tag=ee839502 To: <sip:3000@52.221.122.187>;tag=B4gvey7Ha1e1p Call-ID: mdzZMAwx9AxiY9ax3MxA0A.. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.8.6+git~20190606T190249Z~93b4c92e75~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="52.221.122.187", nonce="4ff129a0-1845-4036-bb10-8f88d20bbdef", algorithm=MD5, qop="auth" Content-Length: 0 recv 813 bytes from udp/[103.50.168.50]:46849 at 07:23:03.165432: ------------------------------------------------------------------------ INVITE sip:3000@52.221.122.187;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.103.30:46849 ;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport Max-Forwards: 70 Contact: <sip:2112@103.50.168.50:46849;transport=UDP> To: <sip:3000@52.221.122.187> From: <sip:2112@52.221.122.187;transport=UDP>;tag=ee839502 Call-ID: mdzZMAwx9AxiY9ax3MxA0A.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rv2.9.27 Allow-Events: presence, kpml, talk Content-Length: 254 v=0 o=Z 1577863381110 1 IN IP4 103.50.168.50 s=Z c=IN IP4 103.50.168.50 t=0 0 m=audio 55560 RTP/AVP 3 101 110 97 8 0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=sendrecv Here i see private ip at Via: SIP/2.0/UDP 192.168.103.30:46849;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport=46849;received=103.50.168.50. Is it any problem? Regards Tanim On Tue, Dec 31, 2019 at 9:56 PM David P <davidswalkabout@gmail.com> wrote:
Search your log for 'Choose rtp'; the IP number following this shows what FS will use for a media stream. There is a line like this for each stream. For your scenario, you want these not to be private IPs.
If they are private, you can try to auto-reject them by changing acl.conf.xml. But you'll have limited success due to this bug https://github.com/signalwire/freeswitch/issues/157 _________________________________________________________________________
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participants (2)
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David P
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Md,Mehedi Hasan Kabir(Tanim)