[asterisk-bugs] [JIRA] (ASTERISK-27885) Calls to SIP phones fail to establish, debug log and tcpdump suggest a packet backlog

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27885?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243599#comment-243599 ] 

Asterisk Team commented on ASTERISK-27885:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> Calls to SIP phones fail to establish, debug log and tcpdump suggest a packet backlog
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27885
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27885
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.17.2, 13.20.0
>         Environment: Debian Stretch: Linux asterisk 4.9.0-4-amd64 #1 SMP Debian 4.9.65-3 (2017-12-03) x86_64 GNU/Linux
>            Reporter: Aragon Gouveia
>            Severity: Minor
>
> The PBX I operate receives calls from a telco via a DAHDI channel, and these typically all ring to SIP phones. At random times, a call will ring to a SIP phone, but answering the phone fails to establish the call. The ringing stops, but audio is just dead.
> I've been running Asterisk with debugging enabled on one SIP extension, while simultaneously running tcpdump in the background to capture the actual network communication. Every time it happens, tcpdump shows the phone's SIP responses reaching the Asterisk server, but the debug log shows the same packets being processed much later than the tcpdump timestamps. By the time Asterisk has processed the phone's response packets, the caller has already hungup or some other timeout has already occurred (depending on which packets get backlogged).
> Unfortunately I have no way to reliably reproduce this. It can take weeks and hundreds of successful calls to manifest. The PBX was recently upgraded from Debian Wheezy + Asterisk 1.8.10.0, and since then this problem started occurring (ie. since Asterisk 13 and Debian Stretch).
> Attaching files with more details:
> core-show-settings.txt - CLI output, "core show settings"
> core-show-sysinfo.txt - CLI output, "core show sysinfo"
> sip-show-settings.txt - CLI output, "sip show settings"
> debug.txt - Debug log, notice the silence between 08:30:07 and 08:30:15
> tcpdump.txt - Notice the delay in processing packets received between 08:30:10 and 08:30:17.



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