The Sip call drops after 30 seconds, but it doesn't always happen. I think it's because of NAT timeout. Am I correct? Or is it something else?
Asked
Active
Viewed 3,667 times
-2
-
1show us some of your code snippets so we can give some idea as in where your sip application is causing problem. – RicoRicochet Dec 23 '14 at 05:28
-
i used default asterisk config – bear Dec 23 '14 at 05:30
-
If you captured the SIP messages on both sides of the call (using a tool like Wireshark), it would probably make the problem clear pretty quickly. – korvus Feb 01 '15 at 05:20
1 Answers
2
It's because 30 seconds is the timeout value for SIP transactions and it's probable that the ACK request, which completes a call INVITE transaction, is not getting through.
As to why the ACK request is not getting through there are a number of possibilities but it's unlikely to be NAT. If it was a NAT issue then the initial INVITE request is unlikely to have reached the callee SIP device.
A common issue can be SIP Application Layer Gateways (ALG) built into home routers. They will often apply very crude text replacements on private IP addresses in SIP packets and this can be enough to break SIP transaction matching logic. Check the specifications on your router model and if it has a SIP ALG turn it off.
sipsorcery
- 30,273
- 24
- 104
- 155
-
i want to try this problem my softphone. but icant config ALG. i configured my client side router: #ip nat inside source list 1 int f0/0 overload #(conf-t)ip nat service sip udp port 8000 #debug ip nat sip am i wrong? – bear Dec 24 '14 at 03:02
-
Were you able to resolve this? I did find this: http://www.phonepower.com/wiki/Linksys_WRT1900AC – billsecond Sep 10 '15 at 00:36