chan_sip: Allow T.38 switch-over when SRTP is in use.

[asterisk/asterisk.git] / channels / chan_sip.c

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c22807c..104ef5c 100644 (file)
--- 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;