[svn-commits] file: trunk r370171 - in /trunk: channels/ include/asterisk/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Jul 18 06:48:08 CDT 2012
Author: file
Date: Wed Jul 18 06:38:05 2012
New Revision: 370171
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=370171
Log:
Fix a crash occurring as a result of excess stack usage.
This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds.
(closes issue ASTERISK-20140)
Reported by: jonnt
Modified:
trunk/channels/chan_sip.c
trunk/include/asterisk/rtp_engine.h
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=370171&r1=370170&r2=370171
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Jul 18 06:38:05 2012
@@ -9367,7 +9367,7 @@
int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;
- struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
+ struct ast_rtp_codecs *newaudiortp = NULL, *newvideortp = NULL, *newtextrtp = NULL;
struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */
struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock();
int newnoncodeccapability;
@@ -9404,10 +9404,11 @@
goto process_sdp_cleanup;
}
- /* Make sure that the codec structures are all cleared out */
- ast_rtp_codecs_payloads_clear(&newaudiortp, NULL);
- ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
- ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
+ if (!(newaudiortp = ast_calloc(1, sizeof(*newaudiortp))) || !(newvideortp = ast_calloc(1, sizeof(*newvideortp))) ||
+ !(newtextrtp = ast_calloc(1, sizeof(*newtextrtp)))) {
+ res = -1;
+ goto process_sdp_cleanup;
+ }
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
@@ -9448,11 +9449,11 @@
if (process_sdp_a_sendonly(value, &sendonly)) {
processed = TRUE;
}
- else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
+ else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec))
processed = TRUE;
- else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
+ else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec))
processed = TRUE;
- else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
+ else if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
processed = TRUE;
else if (process_sdp_a_image(value, p))
processed = TRUE;
@@ -9566,7 +9567,7 @@
ast_verbose("Found RTP audio format %d\n", codec);
}
- ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
+ ast_rtp_codecs_payloads_set_m_type(newaudiortp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9638,7 +9639,7 @@
if (debug) {
ast_verbose("Found RTP video format %d\n", codec);
}
- ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
+ ast_rtp_codecs_payloads_set_m_type(newvideortp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m);
@@ -9702,7 +9703,7 @@
if (debug) {
ast_verbose("Found RTP text format %d\n", codec);
}
- ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
+ ast_rtp_codecs_payloads_set_m_type(newtextrtp, NULL, codec);
}
} else {
ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m);
@@ -9820,7 +9821,7 @@
} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
processed_crypto = TRUE;
processed = TRUE;
- } else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
+ } else if (process_sdp_a_audio(value, p, newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
}
}
@@ -9831,7 +9832,7 @@
} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
processed_crypto = TRUE;
processed = TRUE;
- } else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
+ } else if (process_sdp_a_video(value, p, newvideortp, &last_rtpmap_codec)) {
processed = TRUE;
}
}
@@ -9839,7 +9840,7 @@
else if (text) {
if (process_sdp_a_ice(value, p, p->trtp)) {
processed = TRUE;
- } if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
+ } if (process_sdp_a_text(value, p, newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
processed = TRUE;
} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
processed_crypto = TRUE;
@@ -9912,9 +9913,9 @@
}
/* Now gather all of the codecs that we are asked for: */
- ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
- ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);
+ ast_rtp_codecs_payload_formats(newaudiortp, peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(newvideortp, vpeercapability, &vpeernoncodeccapability);
+ ast_rtp_codecs_payload_formats(newtextrtp, tpeercapability, &tpeernoncodeccapability);
ast_format_cap_append(newpeercapability, peercapability);
ast_format_cap_append(newpeercapability, vpeercapability);
@@ -9977,7 +9978,7 @@
ast_sockaddr_stringify(sa));
}
- ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
+ ast_rtp_codecs_payloads_copy(newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
@@ -10024,7 +10025,7 @@
ast_verbose("Peer video RTP is at port %s\n",
ast_sockaddr_stringify(vsa));
}
- ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+ ast_rtp_codecs_payloads_copy(newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -10048,7 +10049,7 @@
} else {
p->red = 0;
}
- ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
+ ast_rtp_codecs_payloads_copy(newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
} else {
ast_rtp_instance_stop(p->trtp);
if (debug)
@@ -10165,6 +10166,15 @@
process_sdp_cleanup:
if (res) {
offered_media_list_destroy(p);
+ }
+ if (newtextrtp) {
+ ast_free(newtextrtp);
+ }
+ if (newvideortp) {
+ ast_free(newvideortp);
+ }
+ if (newaudiortp) {
+ ast_free(newaudiortp);
}
ast_format_cap_destroy(peercapability);
ast_format_cap_destroy(vpeercapability);
Modified: trunk/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/trunk/include/asterisk/rtp_engine.h?view=diff&rev=370171&r1=370170&r2=370171
==============================================================================
--- trunk/include/asterisk/rtp_engine.h (original)
+++ trunk/include/asterisk/rtp_engine.h Wed Jul 18 06:38:05 2012
@@ -76,7 +76,11 @@
#include "asterisk/res_srtp.h"
/* Maximum number of payloads supported */
-#define AST_RTP_MAX_PT 256
+#if defined(LOW_MEMORY)
+#define AST_RTP_MAX_PT 128
+#else
+#define AST_RTP_MAX_PT 196
+#endif
/* Maximum number of generations */
#define AST_RED_MAX_GENERATION 5
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