Hi everyone,
I’m facing an issue in my VoIP application where there is a noticeable delay in the hangup signal being sent from the UI to Asterisk. Because of this delay, the call recording duration ends up being longer than the actual call duration.
For example, when a user ends the call from the UI, Asterisk receives the hangup signal a few seconds late, and the recording continues until that signal is processed.
My questions are:
What could be causing this delay between the UI hangup action and Asterisk receiving it?
Is this related to WebSocket signaling, SIP timers, or network latency?
How can I ensure that the hangup event is immediately reflected in Asterisk so that the recording duration matches the actual call time?
Any suggestions or debugging steps would be really helpful.
Thanks in advance!
The UI of what? Asterisk doesn’t have an end user interface other than the telephone connections themselves, and the management user interface is not a normal way of ending a call.
More generally, you need to describe the structure of the system that includes Asterisk, as you seem to be assuming lots of detail that don’t form part of Asterisk itself, and you haven’t identified the configuration choices you’ve made on Asterisk. I’m guessing this is SIP VoIP, but Asterisk, can, for example operate as a purely analogue or purely ISDN switch, as well as supporting other VoIP protocols.
The UI is my Voip calling application screen, whenver I click hangup on the screen the call is getting disconnected at the reciever’s phone a bit late sometimes 10 seconds, 14 sec, 40 sec sometimes morethan a Minute, those more than 30 seconds latency we fixed with rtp_timeout and the below 30 seconds latency we need to handle. why was the latency in hangup signal between the UI screen and the recievers side and also it is happenning very rarely among 6k calls some 40 calls like this happening.
You will need to obtain a packet from the machine running Asterisk, e.g. using the CLI command “pjsip set logger on”, and determine the delay between receiving BYE and the call ending. If that delay is reasonably short, you need to pursue this with the developers of the VoIP calling app, or the operators of the network between the two.
The only thing I can think of that might cause delays in Asterisk itself would be a DNS server that is not responding, or responding slowly. However, look in the general Asterisk logging for any processing after the BYE is received.
Telling us what the VoIP calling app is might allow someone to provide help on that.