make RTP handling errors less likely to crash Asterisk (issue #5172)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6584 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index de0ea79..d7e1136 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2660,8 +2660,10 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, char *title)
                if (relaxdtmf)
                        ast_dsp_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
        }
-       tmp->fds[0] = ast_rtp_fd(i->rtp);
-       tmp->fds[1] = ast_rtcp_fd(i->rtp);
+       if (i->rtp) {
+               tmp->fds[0] = ast_rtp_fd(i->rtp);
+               tmp->fds[1] = ast_rtcp_fd(i->rtp);
+       }
        if (i->vrtp) {
                tmp->fds[2] = ast_rtp_fd(i->vrtp);
                tmp->fds[3] = ast_rtcp_fd(i->vrtp);
@@ -2830,6 +2832,12 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
        /* Retrieve audio/etc from channel.  Assumes p->lock is already held. */
        struct ast_frame *f;
        static struct ast_frame null_frame = { AST_FRAME_NULL, };
+       
+       if (!p->rtp) {
+               /* We have no RTP allocated for this channel */
+               return &null_frame;
+       }
+
        switch(ast->fdno) {
        case 0:
                f = ast_rtp_read(p->rtp);       /* RTP Audio */
@@ -2940,8 +2948,8 @@ static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int useg
                p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
                if (videosupport)
                        p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
-               if (!p->rtp) {
-                       ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
+               if (!p->rtp || (videosupport && !p->vrtp)) {
+                       ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", videosupport ? "and video" : "", strerror(errno));
                        ast_mutex_destroy(&p->lock);
                        if (p->chanvars) {
                                ast_variables_destroy(p->chanvars);
@@ -3261,6 +3269,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        int debug=sip_debug_test_pvt(p);
        struct ast_channel *bridgepeer = NULL;
 
+       if (!p->rtp) {
+               ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
+               return -1;
+       }
+
        /* Update our last rtprx when we receive an SDP, too */
        time(&p->lastrtprx);
        time(&p->lastrtptx);
@@ -4316,8 +4329,11 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
                return -1;
        }
        respprep(&resp, p, msg, req);
-       ast_rtp_offered_from_local(p->rtp, 0);
-       add_sdp(&resp, p);
+       if (p->rtp) {
+               ast_rtp_offered_from_local(p->rtp, 0);
+               add_sdp(&resp, p);
+               ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
+       }
        return send_response(p, &resp, retrans, seqno);
 }
 
@@ -4636,7 +4652,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
                        }
                }
        }
-       if (sdp) {
+       if (sdp && p->rtp) {
                ast_rtp_offered_from_local(p->rtp, 1);
                add_sdp(&req, p);
        } else {