Displaying 1 result from an estimated 1 matches for "check_rtp_timeout".
2013 Sep 03
1
Asterisk 11.5.1 / TLS and Media Encryption / Blink as Client / no audio
...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...