[asterisk-bugs] [JIRA] (ASTERISK-18201) Asterisk should fall back to AVP when SRTP module is not loaded and both SAVP and AVP have been offered

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-18201?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan closed ASTERISK-18201.
----------------------------------

    Resolution: Duplicate

Closing this out as a duplicate of ASTERISK-23989. The patch on this issue should be compared with Olle's when it is put up for peer review.

> Asterisk should fall back to AVP when SRTP module is not loaded and both SAVP and AVP have been offered
> -------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-18201
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18201
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.5.0
>         Environment: Asterisk 1.8.5, Nortel NSR
>            Reporter: Fabio Torchetti
>         Attachments: fallback_srtp.patch
>
>
>   When both SRTP and RTP (RTP/AVP and RTP/SAVP) are offered Asterisk should fall back to the RTP protocol if it fails to load the SRTP module. Up until 1.6 Asterisk ignored the SRTP requests and - if available - would fall back to RTP; this is a regression test failure in an environment where SRTP is not to be used, even if it's offered.
>   This is a sample INVITE SDP content of such type:
> v=0
> o=- 189845755 1 IN IP4 XXX.XXX.XXX.XXX
> s=-
> t=0 0
> m=audio 5262 RTP/AVP 0 8 101 111
> c=IN IP4 XXX.XXX.XXX.XXX
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:111 X-nt-inforeq/8000
> a=ptime:20
> a=maxptime:20
> a=sendrecv
> m=audio 5262 RTP/SAVP 8 0 101 111
> c=IN IP4 XXX.XXX.XXX.XXX
> a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:dnUo0FOvQmZF+QJqoT/JlsrcyjFyiDZe5IDM/V                                        6V|2^031|003007014426:004
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:111 X-nt-inforeq/8000
> a=ptime:20
> a=maxptime:20
> a=sendrecv



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