[asterisk-bugs] [JIRA] (ASTERISK-27251) chan_sip doesn't honour rtptimeout setting
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Sep 13 16:03:09 CDT 2017
[ https://issues.asterisk.org/jira/browse/ASTERISK-27251?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-27251:
------------------------------------
Status: Open (was: Triage)
> chan_sip doesn't honour rtptimeout setting
> ------------------------------------------
>
> Key: ASTERISK-27251
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27251
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 13.17.1
> Environment: Ubuntu 16.04 (64-bit)
> Reporter: Ian Gilmour
> Assignee: Unassigned
> Attachments: cli.tgz
>
>
> I have Asterisk running in the cloud with sip.conf configured with:
> {noformat}
> rtptimeout=60
> rtpholdtimeout=300
> rtpkeepalive=20
> {noformat}
> I have a confbridge managed conference room, configured to play MOH if there is only 1 participant.
> I have a SIP client (Jitsi) configured to use TLS+SRTP. It registers with Asterisk on a chan_sip managed IP address + port.
> The SIP client is behind a NAT.
> If the SIP client enters the conference room MOH plays as expected. If I then disconnect the SIP client from the network (so it doesn't deregister, or issue a SIP BYE) I see Asterisk sending media (MOH) to the SIP client's orginal IP address and port indefinitely (>24hours+) and the CLI shows the user as present in the conf call. Media output only terminates if I kick the user manually out of the conf call via the CLI.
> Once the SIP client is disconnected from the network wireshark shows ICMP Destination unreachable messages being returned in response to each Asterisk (MOH) outgoing media packet.
> n.b. I see similar behaviour if I run both Asterisk and client locally. i.e. no NAT.
> The bug seems to be related to ASTERISK-26523.
> Reverting the ASTERISK-26523 change so that it only updates the lastrtprx if the frame isn't AST_FRAME_NULL gives me a chan_sip that honours the sip.conf rtptimeout.
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