Rtp packet transfers after the device internet connection lost during the call

Asterisk Asterisk Support
1

I have configured WebRTC using Pjsip in Asterisk 16 .
During the call ,when one endpoint device goes offline due to internet connection lost, the other end call is not getting hangup ,instead it takes 2 minutes to get hangup.
While debugging with rtp packets using rtp set debug on command, it shows like the rtp packets were transmitted to and from between the two endpoints, even after the internet disconnection of one device.
Any solution for this issue ?
Thanks in Advance :slight_smile:

2

What issue? Normally you want the voice call to be tolerant of network problems.

Do you actually mean lost connection or do you mean one that was actively closed?

There was a recent thread on disconnecting the SIP level when the signalling path was closed at the TCP level without ending the SIP session, which you should look at, although it sounds like the first choice solution, of using RTP timeouts, won’t work for you.

3

The issue is with rtp is getting transmitted even though there was internet connection for the endpoint .
this makes the other end of the call from being hangup ,it waits for 2 minutes untill the rtp transmission stops .Then the call gets hangup.
The thing is whenever the endpoint id not in internet, the call should get disconnected.But, here after internet disconnection of the device also, we could able to see rtp transmission in the log, which makes the other end of the device from disconnecting.

5

also set rtp timeout

rtp_keepalive=10 
                ; Interval, in seconds, between comfort noise RTP packets if
                ; RTP is not flowing. This setting is useful for ensuring that
                ; holes in NATs and firewalls are kept open throughout a call.
rtp_timeout=10
                   ; Hang up channel if RTP is not received for the specified
                   ; number of seconds when the channel is off hold (default:
                   ; "0" or not enabled)
rtp_timeout_hold=300
                   ; Hang up channel if RTP is not received for the specified
                   ; number of seconds when the channel is on hold (default:
                   ; "0" or not enabled)
6

OP says RTP is flowing in both directions so the RTP timeout won’t trip.

7

yes, but often users get confused that they se RTP in both directions on asterisk
but it is just Incomming from A and Outgoing to B
if B is offline due to internet connection lost it cant be sending RTP from B to A
and rtptimeout will detect this and hangup

8

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.