[Freeswitch-users] default muted media on conference ?!

[Stanislav Sinyagin]

probably NAT traversal kicked in?

By default, in SIP profile there's
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>

and auto-nat ACL covers "RFC1918 Excluding your local lan". If you use
only private addresses in your whole ccommunication, it's better to
define an ACL that denies everything and refer to it in "ext-xxx-ip"
parameters.




On Sat, Jul 2, 2016 at 9:29 AM, Anonim Stefan <fanx07 at gmail.com> wrote:
> Hello,
>
> I am trying to establish a conference group call using the setup pictured at
> [1]. Thus, when a user calls to a group extension, all the users in that
> group are called.
>
> I have set the sip-ip and rtp-ip bind IPs. Signaling works fine, the phones
> in the group ring. When I answer I even hear the wave playback conference
> starter/joiner. I can see bindings on the RTP public IP:PORT (using
> netstat). I can ngrep packets coming in and out for that ports.
>
> The problem I have is that I can hear no media.
>
> I've been debugging and eliminated the possible networking firewall issues
> using netcat over UDP and some high ports, between public and private RTP
> ip, in both directions (firewall allows everything).
>
> Thus, I would say that the problem is on the application side. This kind of
> looks like a scenario where all conference users are muted, even if I don't
> have any rules like that in the dialplan group extension. The freeswitch
> ACLs default="allow" and I don't see any "blocked by acl" logs. The logs
> look fine: "... muxing call legs".
>
> Any ideas what might be the problem?
>
> Thanks in advance,
> Stefan
>
>
> [1] https://postimg.org/image/4cwxn91bb/
>
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