chan_sip: Clean leak on error path of process_sdp

Resolve leak in process_sdp that occurs in 2 error path's where
crypto lines are expected but not provided.

ASTERISK-24385 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4045/

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a6038698fcd102f4aa8c45929f35e935ef699c87..fa146e1a5341065b6ec464f11dd633e53e8929b5 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10377,10 +10377,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
                /* Ensure crypto lines are provided where necessary */
                if (audio && secure_audio && !processed_crypto) {
                        ast_log(LOG_WARNING, "Rejecting secure audio stream without encryption details: %s\n", m);
-                       return -1;
+                       res = -1;
+                       goto process_sdp_cleanup_b;
                } else if (video && secure_video && !processed_crypto) {
                        ast_log(LOG_WARNING, "Rejecting secure video stream without encryption details: %s\n", m);
-                       return -1;
+                       res = -1;
+                       goto process_sdp_cleanup_b;
                }
        }
 
@@ -10689,6 +10691,8 @@ process_sdp_cleanup:
        if (res) {
                offered_media_list_destroy(p);
        }
+
+process_sdp_cleanup_b:
        ast_rtp_codecs_payloads_destroy(&newtextrtp);
        ast_rtp_codecs_payloads_destroy(&newvideortp);
        ast_rtp_codecs_payloads_destroy(&newaudiortp);