Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv31357/channels
Modified Files:
chan_sip.c
Log Message:
make RTP handling errors less likely to crash Asterisk (issue #5172)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.850
retrieving revision 1.851
diff -u -d -r1.850 -r1.851
--- chan_sip.c 14 Sep 2005 01:53:06 -0000 1.850
+++ chan_sip.c 14 Sep 2005 02:15:14 -0000 1.851
@@ -2660,8 +2660,10 @@
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 @@
/* 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 @@
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 @@
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 @@
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 @@
}
}
}
- if (sdp) {
+ if (sdp && p->rtp) {
ast_rtp_offered_from_local(p->rtp, 1);
add_sdp(&req, p);
} else {
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