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
........

Merged revisions 429633 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429634 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 7 insertions, 1 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c22807c87..104ef5c51 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10535,6 +10535,12 @@ 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;
@@ -10559,7 +10565,7 @@ 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;