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asterisk
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Fix a crash occurring as a result of excess stack usage.

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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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Joshua Colp 13 years ago
parent 9278b5e51e
commit cbdb2dbb0e
2 changed files with 35 additions and 21 deletions
Show Stats Download Patch File Download Diff File

50
channels/chan_sip.c
Unescape View File

@ -9367,7 +9367,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} 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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} 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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
}
/* 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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} 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)
@ -10166,6 +10167,15 @@ 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);
ast_format_cap_destroy(tpeercapability);

6
include/asterisk/rtp_engine.h
Unescape View File

@ -76,7 +76,11 @@ extern "C" {
#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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