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

[Chris Jones (JIRA)]

Chris Jones created ASTERISK-27609:
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             Summary: 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


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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