[asterisk-users] SIP RTP streams

Da Rock asterisk-users at herveybayaustralia.com.au
Tue Jan 25 07:15:37 CST 2011


In my quest to resolve the ongoing issues I have with outgoing calls...

In the SIP invite, the SDP describes the media and RTP ports. If I run 
sockstat I believe I can see the ports available, but if I run tcpdump I 
see no packets pass or even get blocked at the firewall. How is it 
initiated then? Who starts it?

Cheers



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