Why are SIP calls via my server silent?
- by Archcode
I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually).
I can connect to it, make a call to other endpoint (namely, my android device to my pc and vice versa) and signals are send with no problems (call, answer, hangup, etc). Unfortunately, and what drives me crazy, that's all: no audio gets through, no video either.
Server does not throw errors, it reports many retransmission though, looks like this:
switch_rtp.c:915 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=15
Codecs are set up correctly (same config worked locally on my LAN). NAT/firewall on client side may be a problem, signals do get through (perhaps due to fixed port, data streaming runs on random one, that is currently my best bet). STUN/TURN/ICE setting on client seem to have no effect. Endpoints sit behind symmetric NAT. On server there are no iptables rules, security group is set as suggested there: http://wiki.freeswitch.org/wiki/Firewall
Help, please. How to make it work or at least diagnose what's wrong?
