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-rw-r--r--channels/chan_sip.c50
-rw-r--r--include/asterisk/rtp_engine.h6
2 files changed, 35 insertions, 21 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index cbf88113c..eadf9b961 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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);
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 098530f19..bd47e42b1 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -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
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-08 18:04:54 +0000