Hi all,
In the traditional telephony who owned the call is the caller and is it possible for the caller to change the thorn without lost conversation but after 60sec if the call doesn't live, the central of caller disconnect the call.
My test:
I call from my UA to PSTN and during the call i disconnect my UA from IP connection. in my ethereal i can see now only RTP traffic from PSTN try to reach my UA, but after minutes nothing happen. note that i'm not using an rtpproxy right now. maybe with an rtpproxy it works and disconnect the call.
my question is how can i generate accounting for the call if my UA disconnect the ethernet cable?
the rtpproxy disconnect the call and generate the BYE for the accounting of the call. is it correct? hmm with a b2bua i can solve this issue but if i can do this only with openser is more better.
anyone incurred if this kind of issue?
thanks
tele writes:
I call from my UA to PSTN and during the call i disconnect my UA from IP connection. in my ethereal i can see now only RTP traffic from PSTN try to reach my UA, but after minutes nothing happen.
configure your GW to terminate the call due to missing media packets. if you can't, buy a better GW.
-- juha
But from UA ---> UA there is the same problem.
On Wed, 2006-07-05 at 10:39 +0300, Juha Heinanen wrote:
tele writes:
I call from my UA to PSTN and during the call i disconnect my UA from IP connection. in my ethereal i can see now only RTP traffic from PSTN try to reach my UA, but after minutes nothing happen.
configure your GW to terminate the call due to missing media packets. if you can't, buy a better GW.
-- juha
tele writes:
But from UA ---> UA there is the same problem.
if it is a problem for you, buy a UA that supports session timer. it is not a problem for me, since sip-sip calls don't consume any of my resources once the call has been set up, so why should i care. also mediaproxy detects missing media and closes down its session.
-- juha
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Juha Heinanen -
tele