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

[Mirko Brankovic]

Hi Stefan,

Well, just the know issue that some nat configurations (traversal i think)
may have problems with gathering ice candidates so even when your sip
signaling is ok sdp might be carrying wrong candidates.

But maybe this is not your case, if it works with different media server.

Mirko
On Jul 2, 2016 5:27 PM, "Anonim Stefan" <fanx07 at gmail.com> wrote:

> Hi Mirko,
>
> I will see if I can try the things you proposed.
>
> Could you please tell me what makes you think it's the VPN and what you
> think it might be the misconfiguration in the VPN configuration (e.g.
> "small packets not allowed" etc...)? I see that SIP signaling
> works fine(and advertises the private RTP IP) and I can grep for RTP
> packets coming in and out FS machine, on the RTP public IP.
>
> P.S. In the same configuration, if instead of Freeswitch I have RPTEngine
> (as a media relay) it works fine.
>
> Thank you,
> Stefan
>
> On Sat, Jul 2, 2016 at 11:31 AM, Mirko Brankovic <mirkobrankovic at gmail.com
> > wrote:
>
>> What I think is the problem is the vpn. If you are using sip it is 99%
>> the cause.
>>
>> If you can try without it just as a test. Also try 1on1 call and
>> different uac if possible.
>> Just to eliminate possibilities.
>>
>> Regards,
>> Mirko
>> On Jul 2, 2016 9:30 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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>>
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>>
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>>
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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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>
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