diff options
| author | Joshua Colp <jcolp@digium.com> | 2014-12-16 16:38:34 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2014-12-16 16:38:34 +0000 |
| commit | f26d4618eb7dff4e6c8e8dfd31765bb331bdf2ba (patch) | |
| tree | 819ed0dd74a850156966bd1e165034d339540e0b /channels | |
| parent | ad85e54fd9b545aa6c1dd43a14bdf92344cf4fa0 (diff) | |
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 8 |
1 files changed, 7 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d961730c7..a2604e0c9 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -10526,6 +10526,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action goto process_sdp_cleanup; } + if (p->srtp && p->udptl && udptlportno != -1) { + ast_debug(1, "Terminating SRTP due to T.38 UDPTL\n"); + ast_sdp_srtp_destroy(p->srtp); + p->srtp = NULL; + } + if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, AST_SRTP_CRYPTO_OFFER_OK)))) { ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n"); res = -1; @@ -10550,7 +10556,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action 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; |