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

[George Joseph (JIRA)]

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

George Joseph updated ASTERISK-27609:
-------------------------------------

    Assignee: Chris Jones  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

The new pcap shows that things appear to be normal then twilio sending an encrypted alert (closing) to which asterisk responds with an encrypted alert.  Then twilio closes the TCP connection and asterisk follows suit.  Unfortunately we have no way of knowing WHY twilio sent the alert.  30 seconds earlier, twilio sent a TCP keepalive packet which was ACKed and there were no other errors I can see.  I don't suppose they can provide any logs from their side?


> 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: Chris Jones
>         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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