[Freeswitch-users] No audio for conference call

Wed Jan 1 07:36:53 UTC 2020

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 at 52.221.122.187;transport=UDP>;tag=ee839502

   To: <sip:3000 at 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 at 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 at 103.50.168.50:46849;transport=UDP>

   To: <sip:3000 at 52.221.122.187>

   From: <sip:2112 at 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 at 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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