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

[Chris Jones (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27609?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241881#comment-241881 ] 

Chris Jones commented on ASTERISK-27609:
----------------------------------------

Finally something from Twilio; will keep you in the loop:

I looked into the logs and found in ASTERISK-27609-logger.conf.txt file that your Asterisk server is getting BYE correctly:

    [Jan 22 11:02:17] VERBOSE[813] chan_sip.c:<--- SIP read from TLS:54.172.60.3:35991 --->BYE sip:+18442033202 at 52.91.76.243:5061;transport=tls SIP/2.0CSeq: 14423 BYEFrom: <sip:+12059087858 at asterisk224.pstn.twilio.com;isup-oli=62>;tag=66369999_6772d868_4b1d52f7-a5b4-46ac-af12-9d8a2b01028bTo: <sip:+18442033202 at pbx.statement.services;user=phone>;tag=as482692c8Call-ID: 1b99772d4affe3f5a053c29e43277aef at 0.0.0.0Max-Forwards: 68Via: SIP/2.0/TLS https://urldefense.proofpoint.com/v2/url?u=http-3A __54.172.60.3&d=DwIG-g&c=x_Y1Lz9GyeGp2OvBCa_eow&r=t2YgCTqxzerFRlsCFDyfTv6e6hm4_syTqyTbXV6oNO8&m=I2bUe66XRIMdUN4DmoHZRQ8qMVZ2Kap6oTYtaLeB2-E&s=i_5ngCXtrrRjhAsfvv-9lsXhSHLWPMR8fx_CaxNOUag&e=:5061;branch=z9hG4bKc03c.e98fba07.0Via: SIP/2.0/UDP https://urldefense.proofpoint.com/v2/url?u=http-3A__ 172.18.16.53&d=DwIG-g&c=x_Y1Lz9GyeGp2OvBCa_eow&r=t2YgCTqxzerFRlsCFDyfTv6e6hm4_syTqyTbXV6oNO8&m=I2bUe66XRIMdUN4DmoHZRQ8qMVZ2Kap6oTYtaLeB2-E&s=XFb46VuOPONdh-jw0M27Cam7NcmE3Z81qG-8SzrBHes&e=:5060;rport=5060;received=https://urldefense.proofpoint.com/v2/url?u=http-3A__52.71.97.121&d=DwIG-g&c=x_Y1Lz9GyeGp2OvBCa_eow&r=t2YgCTqxzerFRlsCFDyfTv6e6hm4_syTqyTbXV6oNO8&m=I2bUe66XRIMdUN4DmoHZRQ8qMVZ2Kap6oTYtaLeB2-E&s=XCL-si03hzCX-CeoWqVFcoJY4SBOV9UqsjEwQ66xDbQ&e=;branch=z9hG4bK4b1d52f7-a5b4-46ac-af12-9d8a2b01028b_6772d868_362-11331624058400453305Reason: Q.850;cause=16;text="Terminated"Supported: 100relUser-Agent: Twilio GatewayX-Twilio-CallSid: CA0122c0799eac2ad9ded58f3d7d9da4b4Content-Length: 0

Further investigating I found that this call was terminated from our side, but as it's a single example I'd like to wait until we have a trend.

> 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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