[asterisk-bugs] [JIRA] (ASTERISK-29128) res_srtp: Authentication failure after hold/unhold

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29128?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=253776#comment-253776 ] 

Alexander Traud commented on ASTERISK-29128:
--------------------------------------------

Yeah, that is the SIP standard. Snom does not go further but did not tell me their conclusion. By the way, the ticket-id is [HEU-8749|https://jira.snom.com/servicedesk/customer/portal/2/HEU-8749] now. The Asterisk Team does not have anyone who escalades this to [SIP Implementors|https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors], [SIP Forum|https://en.wikipedia.org/wiki/SIP_Forum], and/or [IETF WG mmusic|https://tools.ietf.org/wg/mmusic/] either.

[~lvl]:
# did you report this to Snom as well (and mention that IDs)? Do you have access to their ticket system at all?
# are you going to drive this further, for example ask the SIP Implementors mailing list?

> res_srtp: Authentication failure after hold/unhold
> --------------------------------------------------
>
>                 Key: ASTERISK-29128
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29128
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 16.13.0
>            Reporter: laszlovl
>            Severity: Major
>              Labels: patch
>         Attachments: filtered.log, snom_changed_srtp_suite_13_256.patch, snom_changed_srtp_suite_13.patch, snom_changed_srtp_suite_16.patch, snom-srtp-debug-filtered.log
>
>
> As simple as the title indicates. Put an SRTP call on hold, unhold it, and Asterisk starts logging "SRTP unprotect failed on SSRC 1509410849 because of authentication failure" afterwards. No more audio is transmitted.
> Traced the problem to commit https://github.com/asterisk/asterisk/commit/c00b032bbfc14f40537989477229f189a1b529d7 (ASTERISK-28903), without it everything works fine.
> Asterisk 16.13, libsrtp 1.5.4.



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