Make sure we don't accept streams we can't handle (bug #3818)
authorMark Spencer <markster@digium.com>
Mon, 21 Mar 2005 15:09:10 +0000 (15:09 +0000)
committerMark Spencer <markster@digium.com>
Mon, 21 Mar 2005 15:09:10 +0000 (15:09 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 152efe5..c005107 100755 (executable)
@@ -2795,8 +2795,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        char host[258];
        char iabuf[INET_ADDRSTRLEN];
        int len = -1;
-       int portno=0;
-       int vportno=0;
+       int portno = -1;
+       int vportno = -1;
        int peercapability, peernoncodeccapability;
        int vpeercapability=0, vpeernoncodeccapability=0;
        struct sockaddr_in sin;
@@ -2839,8 +2839,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        sdpLineNum_iterator_init(&iterator);
        ast_set_flag(p, SIP_NOVIDEO);   
        while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
+               int found = 0;
                if ((sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1) ||
                    (sscanf(m, "audio %d/%d RTP/AVP %n", &x, &y, &len) == 2)) {
+                       found = 1;
                        portno = x;
                        /* Scan through the RTP payload types specified in a "m=" line: */
                        ast_rtp_pt_clear(p->rtp);
@@ -2862,6 +2864,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                        ast_rtp_pt_clear(p->vrtp);  /* Must be cleared in case no m=video line exists */
 
                if (p->vrtp && (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
+                       found = 1;
                        ast_clear_flag(p, SIP_NOVIDEO); 
                        vportno = x;
                        /* Scan through the RTP payload types specified in a "m=" line: */
@@ -2879,6 +2882,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                                while(*codecs && (*codecs < 33)) codecs++;
                        }
                }
+               if (!found )
+                       ast_log(LOG_WARNING, "Unknown SDP media type in offer: %s\n", m);
+       }
+       if (portno == -1 && vportno == -1) {
+               /* No acceptable offer found in SDP */
+               return -2;
        }
        /* Check for Media-description-level-address for audio */
        if (pedanticsipchecking) {
@@ -2938,21 +2947,21 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
         */
        sdpLineNum_iterator_init(&iterator);
        while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
-      char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
-         if (!strcasecmp(a, "sendonly")) {
-               sendonly=1;
-               continue;
-         }
-         if (!strcasecmp(a, "sendrecv")) {
-               sendonly=0;
-         }
-         if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
-         if (debug)
-               ast_verbose("Found description format %s\n", mimeSubtype);
-         /* Note: should really look at the 'freq' and '#chans' params too */
-         ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
-         if (p->vrtp)
-                 ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype);
+               char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
+               if (!strcasecmp(a, "sendonly")) {
+                       sendonly=1;
+                       continue;
+               }
+               if (!strcasecmp(a, "sendrecv")) {
+                       sendonly=0;
+               }
+               if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
+               if (debug)
+                       ast_verbose("Found description format %s\n", mimeSubtype);
+               /* Note: should really look at the 'freq' and '#chans' params too */
+               ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
+               if (p->vrtp)
+                       ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype);
        }
 
        /* Now gather all of the codecs that were asked for: */
@@ -8273,8 +8282,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                if (p->owner) {
                        /* Handle SDP here if we already have an owner */
                        if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
-                               if (process_sdp(p, req))
+                               if (process_sdp(p, req)) {
+                                       transmit_response(p, "488 Not acceptable here", req);
+                                       ast_set_flag(p, SIP_NEEDDESTROY);       
                                        return -1;
+                               }
                        } else {
                                p->jointcapability = p->capability;
                                ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");
@@ -8298,8 +8310,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                }
                /* Process the SDP portion */
                if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
-                       if (process_sdp(p, req))
+                       if (process_sdp(p, req)) {
+                               transmit_response(p, "488 Not acceptable here", req);
+                               ast_set_flag(p, SIP_NEEDDESTROY);       
                                return -1;
+                       }
                } else {
                        p->jointcapability = p->capability;
                        ast_log(LOG_DEBUG, "Hm....  No sdp for the moment\n");