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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
This commit is contained in:
Kevin P. Fleming 2005-09-14 02:15:14 +00:00
parent 811ec06cfd
commit e60671d5bf
1 changed files with 23 additions and 7 deletions
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30
channels/chan_sip.c
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@ -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 {