Fix a crash occurring as a result of excess stack usage.
[asterisk/asterisk.git] / channels / chan_sip.c
index cbf8811..eadf9b9 100644 (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);