[asterisk-bugs] [JIRA] (ASTERISK-26003) Unexpected call termination when using PJSIP + SRTP

[Ian Gilmour (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26003?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Ian Gilmour updated ASTERISK-26003:
-----------------------------------


Problem traced to a misconfigured OpenSIPS server sitting between SIP client and Asterisk. This SIP server was failing to forward the client SIP UPDATE pkts (which contained the session timer update) on to Asterisk. The SIP client was requesting a 1800 sec session expiry, as a result of which Asterisk (PJSIP sip_timer.c) was setting up a timeout of 1800-32=1768secs. Because the session update never got received Asterisk correctly terminated the call after 29mins 28sec.

> Unexpected call termination when using PJSIP + SRTP
> ---------------------------------------------------
>
>                 Key: ASTERISK-26003
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26003
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.7.2
>         Environment: centos 6 (64-bit)
> asterisk 13.7.2
> pjproject 2.4.5
> libsrtp 1.5.2
>            Reporter: Ian Gilmour
>            Assignee: Ian Gilmour
>
> Hi, I'm seeing calls consistently terminating ~29-30mins into the call (say to the Asterisk echo()) when using chan_pjsip (with TLS) and SRTP (media_encryption=sdes). If I change pjsip.conf to use RTP (media_encryption=no) it all works fine.
> Should I be using another version of libsrtp? Or some other combination of source code packages?



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