[asterisk-users] Asterisk 11.5.1 / TLS and Media Encryption / Blink as Client / no audio

[Thorsten Göllner]

Hi,

I use Asterisk 11.5.1 and it works fine. :)

Now I want to use TLS and media encryption. I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial

When I place a call via Blink-Client (0.5.0) I get connected and Blink 
shows 2 locks. The blue lock shows "Signaling is encrypted using TLS" 
and the orange lock shows "Media is encrypted using sRTP". BUT i hear no 
audio. After ~60 seconds I get the following message:
NOTICE[21005]: chan_sip.c:28800 check_rtp_timeout: Disconnecting call 
'SIP/tgoellner-0000002c' for lack of RTP activity in 62 seconds

"sip show peers" shows me, that my Blink-Client is registered on port 
60071. All other SIP-Clients (no TLS an no media encryption) are 
registered at port 5060.

I tried to open the tcp and udp port range from 10000 to 61000 (in 
iptables). But with no success.

I am not sure, but I think it's a firewall/NAT problem?! (Yes, my client 
is behind a router > NAT)

Any idea?

-Thorsten-