[asterisk-users] No RTP from asterisk?

[Duncan Turnbull]

On 1/03/2010, at 2:41 PM, Peter Serwe wrote:

> I checked the firewall, iptables -L showed no rules whatsoever.  No other traffic has indicated it was blocked, iptables was set in allow all everywhere mode.
> 
> I went ahead and turned it off, still don't have RTP.  No audio either direction via lines registered.
> 
> G729 is completely disabled from all trunk groups and users, only using G711 at this point.
> 
> Peter
> 
> 
The asterisk rtp debug should show if asterisk is sending audio or receiving packets but its not nearly as useful as tcpdump. 

tcpdump udp port 5060 -s0 -A will give you all the SIP. 

But just dumping all traffic between asterisk and the host will give you a view on RTP - you should see it take off when a call is setup if its not blocked

You should see a SIP Invite to setup a call with the audio destination - this should be your asterisk box and the far end depending on who is doing what. You should look at the address and also whether both sides are providing a mutually acceptable audio formats. If there are no agreed audio formats you won't get rtp. The c= in the session setup indicates the addresses each site is using for media. 

Cheers Duncan