[asterisk-bugs] [JIRA] (ASTERISK-27609) sip_tcptls_read: SIP TCP/TLS server has shut down after 120s

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27609?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-27609:
-------------------------------------

    Status: Waiting for Feedback  (was: Waiting for Feedback)

> sip_tcptls_read: SIP TCP/TLS server has shut down after 120s
> ------------------------------------------------------------
>
>                 Key: ASTERISK-27609
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27609
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/TCP-TLS
>    Affects Versions: 15.1.3
>         Environment: AWS EC2 running Centos 7
>            Reporter: Chris Jones
>            Assignee: Asterisk Team
>         Attachments: asterisk-27609.core.tar.gz, asterisk-27609.pcap.tar.gz, asterisk-27609.tcpdump, ASTERISK-27609.txt, asterisk-cli-output.txt
>
>
> I have a SIP provider (Twilio) that has provided working guidelines for configuring Asterisk to have a Secure SIP tunnel between it and Twilio. I have implemented the TLS and SRTP, and configured Twilio to be Secure-only.
> I can make and receive calls just fine - TLS and SRTP is working. 
> If I hang up the call within 2 minutes: the call disconnects perfectly fine on both ends.
> If the call is longer than two minutes,
>  - at exactly the 120s mark, I get the following sip debug:
> DEBUG[30015]: iostream.c:157 iostream_read: TLS clean shutdown alert reading data
> DEBUG[30015]: chan_sip.c:2905 sip_tcptls_read: SIP TCP/TLS server has shut down
>   - if I hang up the call on the Asterisk side, my mobile phone that dialed into Twilio, does not hang up. 
>   - after 300seconds beyond the initial 120+ seconds, my mobile phone hangs up. I feel this a Twilio timeout. Interesting enough, Twilio sends a BYE but Asterisk responds that the call leg does not exist anymore.
> Because call setup and SRTP both work, the hangup issue is an aggravation because it could increase toll charges (extra 5 minutes every call if the other doesnt hang up). And we need to resolved before we can deploy our app.
>  



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