Thanks for the reply. Got your point about NATing issue and no response of 200 OK and as a resoult ACK Timeout. So, now to resolve the issue, if you can assist, what could be the possible fixies?
From where can i start? where to look?
Thanks. -- Thanks, Rutu Patel <http://www.inextrix.com> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi@avimarcus.net> wrote:
5 second response: 32 seconds is a timer/[network/NAT] issue.
You have lots of 200s to the user since it's waiting for an ACK and keeps retrying, but for whatever network reason (router... sip alg?), it isn't getting one, so it triggers a timer to stop the call.
-Avi Marcus BestFone
On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel@inextrix.com> wrote:
Hello,
Having issue of call dropping after 32 seconds, here are the details-
x.x.x.174: opensips server x.x.x.166: freeswitch server x.x.x.3: another opensips server which is registered as gateway on above freeswitch server x.x.x.6: freeswitch server x.x.x.47: server through which the user is registered I am trying to call from xxxx9 to xxxxxxx29858 xxxxxxx00181 is caller-id name and caller-id number
Call flow is like this: registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
Outbound-proxy is set to x.x.x.174 in Gateway configuration.
1) Call is initiated by the user(xxxx9) registered with host x.x.x.174 2) Call hit the freeswitch server x.x.x.166 3) After '180 Ringing' and '183 Session Progress' packet sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the gateway x.x.x.3 But after 32 seconds call is dropped, Within 32 seconds audio is ok from both end so it should not be the RTP issue. Here I have attached the file with sip logs, you can observer from the file that, there are many '200 OK' from x.x.x.174 to x.x.x.166 and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
What is wrong here? Any help would be appreciated here.
Here is the file with sip logs -- Thanks, Rutu Patel <http://www.inextrix.com>
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