Author: bebuild Date: Fri Dec 19 14:34:34 2014 New Revision: 429857 URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=429857 Log: chan_sip: Allow T.38 switch-over when SRTP is in use.
Previously when SRTP was enabled on a channel it was not possible to switch to T.38 as no crypto attributes would be present. This change makes it so it is now possible. If a T.38 re-invite comes in SRTP is terminated since in practice you can't encrypt a UDPTL stream. Now... if we were doing T.38 over RTP (which does exist) then we'd have a chance but almost nobody does that so here we are. ASTERISK-24449 #close Reported by: Andreas Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523) ........ Merged revisions 429632 from http://svn.asterisk.org/svn/asterisk/branches/11 Modified: certified/branches/11.6/ (props changed) certified/branches/11.6/channels/chan_sip.c Propchange: certified/branches/11.6/ ------------------------------------------------------------------------------ Binary property 'branch-11-merged' - no diff available. Modified: certified/branches/11.6/channels/chan_sip.c URL: http://svnview.digium.com/svn/asterisk/certified/branches/11.6/channels/chan_sip.c?view=diff&rev=429857&r1=429856&r2=429857 ============================================================================== --- certified/branches/11.6/channels/chan_sip.c (original) +++ certified/branches/11.6/channels/chan_sip.c Fri Dec 19 14:34:34 2014 @@ -10356,6 +10356,12 @@ goto process_sdp_cleanup; } + if (p->srtp && p->udptl && udptlportno != -1) { + ast_debug(1, "Terminating SRTP due to T.38 UDPTL\n"); + sip_srtp_destroy(p->srtp); + p->srtp = NULL; + } + if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) { ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n"); res = -1; @@ -10380,7 +10386,7 @@ goto process_sdp_cleanup; } - if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) { + if (!(secure_audio || secure_video || (p->udptl && udptlportno != -1)) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) { ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n"); res = -1; goto process_sdp_cleanup; -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- svn-commits mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/svn-commits
