[SIP Beyond VoIP] STUN keep-alive

Adrian Georgescu ag at ag-projects.com
Sun Jan 15 20:27:25 CET 2012


Your SIP server must be able to keep the NAT open. See OpenSIPS nat traversal module as an example, each SIP server has a similar mechanism. Without support from the server you you will not be able to do it.

Adrian 
 
On Jan 15, 2012, at 8:15 PM, Mihai Richard wrote:

> Hello,
> 
> I am having some problems with starting an audio call between two clients if they registered at some time one from another(aprox. 1 minute). I used ngrep to see sip traffic and this is what I saw: if clients register at aprox. the same time(less then a minute apart) if one of them sends the INVITE message the other client receives it, however if they’ve registered at some time one from another, the first client to have registered does not receive the INVITE. With ngrep on the server I see that the server receives the INVITE from one client and attempts to send it to the other client several times but without success.
> 
> My guess is that this happens because the route created through the NAT by the client not receiving the INVITE is closed due to inactivity. My question is how could I send some messages to the server and keep that route open?
> 
> Also if I try to start an audio call from the client that didn’t receive the INVITE, the call starts normally but on the server I see that the packets containing the INVITE message have a different source port than the packets containing the REGISTER messages from the same client.
> 
> Thank you,
> Mihai Richard
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