[asterisk-bugs] [JIRA] (ASTERISK-16898) SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-16898?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226103#comment-226103 ] 

Alexander Traud commented on ASTERISK-16898:
--------------------------------------------

Thanks to [the community|https://github.com/cisco/libsrtp/issues/29], I was able to enhance the filter, so all Nokia Symbian/S60 are recognized now. Furthermore, I added/updated the patches for the latest releases (Asterisk 13, libsrtp 1.5, and libsrtp2-dev).

[~mjordan] because I was not able to find any related changes on libsrtp/GitHub, yet, which fixes are you about? The Asterisk patches require a switch/patch within the libsrtp library. That patch is attached here to this issue, but is not included in libsrtp. It got rejected. However, because I did not understand why, I was not able improve the patch and/or persuade the libsrtp maintainers further. If somebody is interested in these patches (which are about Nokia Symbian/S60 devices only), please, take over and continue!

Until then, we could incorporate parts of the debug patch into Asterisk: For example with Nokia Series 40 (and many other platforms), that unprotect warning happens not for audio/voice (RTP) but statistical (RTCP) messages. With the more precise warning from the debug patch, this might help others to investigate that (different) issue/cause further. Interested? Then, I spin-off that change about the warning message(s) and go through review.

> SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0
> ----------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-16898
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-16898
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/SRTP
>            Reporter: Marcello Ceschia
>         Attachments: srtpROC-Debug_for_Asterisk11.patch, srtpROC-LE_for_Asterisk11.patch, srtpROC-LE_for_Asterisk13.patch, srtpROC-LE_for_libsrtp1.4.5.patch, srtpROC-LE_for_libsrtp1.5.x.patch, srtpROC-LE_for_libsrtp2.patch
>
>
> {noformat}
> [Jan 19 09:29:34] WARNING[9825] res_srtp.c: SRTP unprotect: authentication failure
> {noformat}
> {noformat}
> Useragent    : Nokia RM-530 052.005
> Prim.Transp. : TLS
> {noformat}
> *ADDITIONAL INFORMATION*
> Depending on the initialze sequence number, the audio stream will become one way.
> I did several tests with the same result always.



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